mp4tools are a set of scripts to encode Audio and Video in Isomedia (aka MP4) format
mp4tools uses opensources programs as backends like mencoder, MP4Box, xvid, x264
and its quality is very good.
There are presets for most common devices like Sony PSP, Apple iPod, Nokia S60 phones,
so you haven't to go in the jungle of encoding options, firmware bugs, unsupported configurations
common in all embedded devices.
mp4tools hot functions are:
After installing you will have two type of scripts, some named mk* and other named dvd2*
For example, mkipod encodes a video file to the ipod format, while dvd2ipod rips a DVD into the same format
The included scripts are:
Additionally, there are amrenc and aacplusenc, which respectively encodes to AMR and AAC+
Paul2008-09-01 03:27:48
Hi, I love these tools, makes everything so easy. The only problem I have is when using your aacplusenc 0.17.1 the resulting files have corrupted audio. There is distortion when playing it on my phone, and mplayer shows errors when seeking when i play it on my PC. When using neroAacEnc the audio is fine and mplayer does not show any errors (but it is slower and the resulting files are larger). I'm using 64-bit linux, maybe that has something to do with it?
diverbelow2008-08-27 14:41:27
I was getting the same output as @Acesabe, using mk3gp, then I tried mks60 and I got the converted 3gp file I needed.
Acesabe2008-08-25 16:04:50
The script works (tried mk3gp and dvd23gp) but there is never a resulting file saved anywhere.
Also get Mplayer error see below - not sure why...
>>>>
Ripping Audio
MPlayer 1.0rc1-4.1.2-DFSG-free (C) 2000-2006 MPlayer Team
CPU: AMD Athlon 64 Processor 3000+ (Family: 15, Model: 4, Stepping: 10)
CPUflags: MMX: 1 MMX2: 1 3DNow: 1 3DNow2: 1 SSE: 1 SSE2: 1
Compiled with runtime CPU detection.
mplayer: could not connect to socket
mplayer: No such file or directory
A: 17.3 V: 8.3 A-V: 8.990 ct: -0.032 50/ 50 0% 0% 0.9% 50 0
************************************************
**** Your system is too SLOW to play this! ****
************************************************
Possible reasons, problems, workarounds:
- Most common: broken/buggy _audio_ driver
- Try -ao sdl or use the OSS emulation of ALSA.
- Experiment with different values for -autosync, 30 is a good start.
- Slow video output
- Try a different -vo driver (-vo help for a list) or try -framedrop!
- Slow CPU
- Don't try to play a big DVD/DivX on a slow CPU! Try some of the lavdopts,
e.g. -vfm ffmpeg -lavdopts lowres=1:fast:skiploopfilter=all.
- Broken file
- Try various combinations of -nobps -ni -forceidx -mc 0.
- Slow media (NFS/SMB mounts, DVD, VCD etc)
- Try -cache 8192.
- Are you using -cache to play a non-interleaved AVI file?
- Try -nocache.
Read DOCS/HTML/en/video.html for tuning/speedup tips.
If none of this helps you, read DOCS/HTML/en/bugreports.html.
A: 793.7 V: 793.7 A-V: -0.028 ct: 0.207 11832/11832 0% 0% 0.4% 11033 0
galwa2008-08-18 10:41:10
thanx,
re entrant -> The option to open multiple encoders. I guess it requires the removal of all static declarations for context approach.
teknoraver2008-08-17 13:33:04
it supports PCM input as:
mono or stereo
32, 44.1 or 48 khz
16 bit signed
PS is enabled on low bitrates (check the source) while SBR is mandatory
re entrant?
galwa2008-08-14 12:29:09
Hay, I'm trying to incorporate thee aacplusenc into a project. some questions regarding it.
what PCM/RAW input bit rates does the encoder supports? (or more to the point what bitrate/channels/sampelrate combinations)
Should it be possible to use the encoder for AAC-HE v1 (does the PS mandatory)?
should it be possible to use the encoder for AAC only (no sbr)?
Do you have plans to add support for re entrant ?
teknoraver2008-08-06 21:41:49
I know, i'll fix it when i'll have the time
i'm busy at work now
see you :)
Joel2008-08-03 13:49:35
trying to run dvd2psp I get:
Encoding H.264 Video (1st pass)
...
The ofps option must be a floating point number or a ratio (numerator[:/]denominator): -ofps
Error parsing option on the command line: -ofps
Then says the same thing for pass 2 and then exits.
Any suggestions (PS I have also tried commenting out line 232 in mp4tools as per Randy below with no effect)
pablo2008-07-24 22:38:39
cool :)
teknoraver2008-07-22 01:48:08
@NoSmile
maximum bitrate is 64kbit.
AAC+ achieves the same quality of AAC at half the bitrate,
so there is no need t ogo above 64kbit.
If you want higher bitrates, use plain AAC
NoSmile2008-07-19 15:28:16
Hi, I tried aacplusenc on Leopard (10.5.3) compiled myself with apple's compiler and it works fine.
There is just one problem : I can't encode more than 64kbps and I'd like to be able to encode on 80 kbps...
Another question : Is there any guidelines on using the libraries? I'd like to use them to build a stream encoder.
Thanks
teknoraver2008-07-17 15:16:45
@Rommel
hehe i know, mkipod encodes as x264 while mks60 encodes with xvid.
You get better video but worse audio as mkipod uses AAC while mks60 uses AAC+
Edit your mks60 to use x264 and AAC+ and you'll have perfect settings.
I can't enable x264 in mks60 by default because some phones with less powerfull CPUs will not play it (eg. E65)
Rommel2008-07-11 14:58:00
Works GREAT!!!! in my N95
Use mkipod which give superior quality over mks60
Thanks!!!!!
dizzy2008-07-10 03:31:24
I have a situation in which I have been given AAC+ raw. I would like to convert it to AAC with ADTS - seems like an approach that would work with most embedded players. What do you think?
ItaloMaia2008-07-07 01:11:02
Fair enough, and good to know.
Well, i kind of need the mks60 to accept custom width, height and quality. My s60, per example, only accepts mks60 output if it has lower width/height. Oh, and mks60 goes bad with some filename. This one "[AuN-Bakadeshi] Let's just be SUPER CUTE GIRLS! (704x396 Xvid).avi" and a few others where pretty trick for my. I had to give a output name for it to work.
teknoraver2008-07-04 01:45:25
Italomaia:
mp4tools are in medibuntu: http://medibuntu.org/
teknoraver2008-07-04 01:44:36
dizzy:
Why convert aac+ to aac?
Just encode to plain aac!
dizzy2008-07-02 23:00:23
Hey,
Do your tools convert AAC+ to AAC with ADTS? Seems that embedded players don't like AAC+ yet.
Italomaia2008-07-01 05:34:38
I changed width to 192 and height to 144
and them the mks60 output was up and running in real media for my nokia 6120. Thanks and if you would like to make a interface or/and change your program to python, i would gladly help! This babe should be in ubuntu repository, for sure.
Big hug.
italomaia2008-06-30 09:17:09
The program could no read the audio format.
italomaia2008-06-30 09:16:43
mks60 output vids where not compatible with coreplayer 1.2 :/
Stephen2008-06-27 18:31:38
hello, ive got a chinese phone Dual Sim ,Touch screen, It has some deafult 3gp video files which it plays well..I tried to convert some mpg files to 3gp n copied it to the memory card but it doesnt play sayin invalid file format..
Can some body help
Thank You..
teknoraver2008-06-17 15:04:32
Comments are back ;)
cihun2008-05-12 06:59:48
i like it!
junior2008-04-19 13:46:45
Just migrated to Hardy. And I've read that it don't support bash scripts no more, cause of the new dash (don't know much about this) But will there be mp4tools-packages for Hardy soon? I miss your program :)
Thomas2008-04-17 14:58:49
Could you possibly post a sample command line syntax?
I can't seem to figure out what the program wants, i.e. where does it look for the input file?
Thanks for the help!
Randy Perkins2008-04-17 06:17:31
let me try again
great scripts, thanks
as far as the problem lucas is having
i think i figured it out.
i am using the ubuntu build from your repository.
in mp4tools. line 149 , the variable $OFPS has the switch -ofps added to it. this makes the encoding fail because on line 232 -ofps is prepended to the variable again.
i tried to post a diff earlier but it didnt work
i have a verizon voyager and using dvd2s60 and mk360 works for me , i am saving my files with a .mp4 extension.
additionally sometimes the crop detect results in heights and widths that will not play properly on my phone. I added 'exact' as the 5th paramater to the scaleres call within dvd2s60. I'm sure this issue varies from phone model to model
randy2008-04-16 19:18:33
my message didnt post and it wasnt spam
teknoraver2008-04-16 16:50:01
no spam please
teknoraver2008-04-12 00:23:08
@lucas
try to echo the mencoder command given by mp4tools, and run it as:
DEBUG=1 dvd2ipod testdvd.m4v
Anaspeople2008-04-11 23:17:01
Great job! Thanks a lot for sharing it with the world!
Genaro2008-04-11 04:22:48
How do i see the mp4 on the psp?
lucas2008-04-10 23:22:49
hi, i do not understand what i am doing wrong?!
:~$ dvd2ipod testdvd.m4v
... everything looks ok but then:
Encoding H.264 Video (2nd pass)
MEncoder 2:1.0~rc1-0ubuntu13.2+medibuntu1 (C) 2000-2006 MPlayer Team
CPU: Intel(R) Core(TM)2 Duo CPU E6750 @ 2.66GHz (Family: 6, Model: 15, Stepping: 11)
CPUflags: Type: 6 MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 1
Compiled with runtime CPU detection.
The ofps option must be a floating point number or a ratio (numerator[:/]denominator): -ofps
Error parsing option on the command line: -ofps
Exiting... (error parsing command line)
Muxing
Option unknown. Please check usage
saved to
Option unknown. Please check usage
Fraser2008-04-05 18:52:01
Yet another question:
how do i change the bitrate of aac?
Fraser2008-04-05 18:39:48
Thanks a lot.
Great work!
teknoraver2008-04-05 18:21:02
@Fraser
AQ is teh audio quality: 0 is lowest and 1 maximum
Fraser2008-04-05 17:46:35
Great results!!!
But is there a possibility to increase the audio quality?
What does AQ='0.2' do?
Thanks
fraser
Nic2008-04-03 01:35:50
Did someone try this on PPC?
I'm on Leopard PPC and tried with both compilers, Apple's and GCC4.2.3.
Both compiled w/o error but files soud like you have a really bad radio reception.
I have an older version of your encoder(0.7) that works fine so it might be an endian issue somewhere in the new code!?
teknoraver2008-04-01 01:18:41
@johnnyf
sorry, there are linux scripts
teknoraver2008-03-29 17:41:24
@peal
There are some errors in your script:
1) nokiatagger works on MP4 files, not AAC ones
2) you don't wrap the AAC file into an MP4
3) you don't use normalize-audio
4) mkmp4 does it all
Renato2008-03-28 22:15:13
Hi,
I'm a total newbie so it might be that the solution to this problem is easy to find... but I just don't know where to look for.
I converted a .flv video using mkipod but I have an audio delay of about 3 secs. How can I solve the problem?
Thanks
peal2008-03-28 18:12:06
Hi,
I just rewrote someone's script to convert all mp3s in a folder to aacplus. Thought this might be wanted by you guys so I post it here.
/peal
http://pastebin.ca/962217*edit, please don't post huge comments
micho362008-03-23 22:30:26
gdzie jest mp4 tools
Charles Goyard2008-03-06 16:10:46
Hi, thanks for your answer. Maybe you should consider converting your "view" video samples to something the flash player can read ? -- Regards
geierb2008-03-03 19:50:36
I do audio streaming with aacplusenc and an apache cgi script (my upstream is only 64kbit/s).
I just send the right header to the client (Content-Type: audio/3gpp) and convert my audio files to aac on the fly.
Here's an example for re-encoding mp3 files:
madplay -Q -b 16 -R 44100 -o wave:- \"$mp3filename\" | aacplusenc - - 32000 s 2>/dev/null
Basically madplay converts the mp3 to wav and writes it to stdout, aacplusenc picks it up with stdin, converts it and writes the aac data back to stdout. There apache picks it up automatically and streams it to the client.
Maybe this helps you with DSS.
teknoraver2008-03-03 13:54:23
these scripts can encode, but dunno how to stream
Joost2008-03-02 23:06:09
Great collection of tools, many thanx!
Would it be possible with any of these tools to encode a live stream on the fly and hint it to DSS (Darwin / stream.dsp) (with some modifications maybe). That would be more then a great feature...
Anyone got some hints on this.. i played with vlc, but it's not working very well (at all ;)).
Cheers
teknoraver2008-02-27 15:44:31
flash player can't play movies muxed with -sbrx, only -sbr, so...
Charles Goyard2008-02-27 15:30:19
Hi,
I can't get the sound of your samples to work with the flash player (0.9.115) under Linux/Firefox. With samples from other sites it's ok. How come ?
Regards,
teknoraver2008-02-27 15:24:33
for i in dir/* ; do
mkmp4 "$i"
done
junior2008-02-27 15:08:30
never mind... made a bash script to convert a folder at the time
junior2008-02-27 11:08:02
But hey! How do I convert a folder at the time? It's really annoying to use the terminal for one file at the time
Alec2008-02-27 08:51:18
2 teknoraver:
Thanks for response, I found how to switch scaling off on ipod in settings (I'm just new to ipod, got it 3 days ago), sorry for confusion. Cropped video is shown as it should be now.
oscord2008-02-27 08:14:12
Hey!
i've run into a problem compiling aacenc. Can't build aacenc on RHEL4 u5 32 bit (Intel). I get a lot of warnings and error while trying to run "make install" http://pastebin.com/m3610901
Please let me know if I'm missing any other packages.
Grazie.
Oscord.
teknoraver2008-02-26 23:54:07
@junior
if the MP3 is good (eg. 128+ bitrate) the loss isn't noticeable
for the mplayer warnings, it writes many, but they are harmless
junior2008-02-26 22:43:01
Ok, have a couple of more questions: I get this error:
Ripping Audio
MPlayer 1.0rc2-4.1.3 (C) 2000-2007 MPlayer Team
CPU: Genuine Intel(R) CPU T2250 @ 1.73GHz (Family: 6, Model: 14, Stepping: 8)
CPUflags: MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 1
Compiled with runtime CPU detection.
mplayer: could not connect to socket
mplayer: No such file or directory
Does it matter? I get a nice output of very good quality! :)
And how do I transcode a bash of files?
junior2008-02-26 22:10:40
I missed build essentials earlier.. But now I've got it all. One question; Do I need to have a wav to convert to aac+? Or do I get the same quality if I use mks60 when I already have an mp3?
teknoraver2008-02-26 22:06:53
authentication? you mean the gpg key? no
Joel2008-02-26 21:59:03
do you have an authentication for your repo?
teknoraver2008-02-26 20:09:45
@Alec
try using cropres instead of scaleres in mkipod
Alec2008-02-26 14:26:18
On Fedora 8, "normalize-audio" binary is just "normalize", would be also good to autodetect ...
Alec2008-02-26 14:01:49
The 16:9 video for ipod is scaled by ipod itself, thus all videos are shown as 4:3 cropped.
I wonder if it is possible to add a parameter to the scripts in order not to remove the letterbox on 16:9 videos or to add a missing one, something like:
"mkipod -letterbox <input>" ...
All the rest is perfect, many thx to the developers.
teknoraver2008-02-25 00:21:15
btw, try to compile it now, i changed it a bit
teknoraver2008-02-24 23:59:15
If size_t is undeclared then your compiler has serious issues
Junior2008-02-24 22:48:36
Eh, how exactly do you use the nokiatagger? I only get errors when I enter "gcc nokiatagger.c -o nokiatagger", ex "184: error: ‘size_t’ undeclared (first use in this function)"
Chris2008-02-23 05:43:57
re: threads=auto
thanks!
Sebastian2008-02-13 14:30:34
Thanks for the tools, I downloaded the aacplus encoder (win binary) and works great!
teknoraver2008-02-11 03:03:07
I'm not a Mac expert, I compiled it on my dad mac (Leopard) and seemed to work.
But I used a decent compiler (GCC4.2.3) instead of the crappy Apple one (GCC 4.0.0)
Simon2008-02-10 23:52:13
Hi, I compiled the 0.16 aacplusenc version on mac osx tiger (10.4.11). Everything was nice except I had to remove #include <endian.h> cause tiger does not have any endian.h. (Compilation seem fine although)
You maybe had only tested it on Leopard?
The source version run ok but produce crasy files. Sound seem downsampled and music is NOT present.
Finally, running the precompiled version on my tiger lead to a beautifuel Bus Error...
sarg2008-02-01 09:37:00
thanks, new release works fine
teknoraver2008-02-01 02:50:09
@Andy
for LC-AAC there is faac.
If you want lower bitrates do a mono encoding, a 22050 Hz one, or both
Andy2008-01-31 21:33:25
aacplusenc is an excellent encoder - thanks. Is there any chance of an option to encode in LC-AAC? (I know this is not in the spirit of the "aacplus" name but my ipod ignores the SBR+PS so is mono and half sample rate - quality still excellent even then but I need low bit rate for some audio books.)
teknoraver2008-01-31 19:40:26
@Chris
threads=auto works with new x264 in hardy, i'll add it
teknoraver2008-01-31 19:11:47
@sarg
rawvideo is needed.
Also the scaling options are in $VF
add a line:
echo "Video Filters where $VF"
*after* the encode
teknoraver2008-01-31 19:08:06
@CaptnBlack
I haven't tried the windows build so much, however the stdin thing is standard C (fread(stdin, ...))
dunno how could it be
sarg2008-01-30 17:04:01
mkpsp has been broken in new release
i'm think bug is here:
mencoder "$1" $VF $OFPS -nosound -o "$2" ---> -of rawvideo <---
with that option mkpsp don't scaling video for appropriate dimensions (this option overrides scaling in $VF)
and another bug: (maybe only in my bash (3.2.25))
mkpsp exits after message "Scaling a 656x352 movie to 480x272"
i'm think that's becouse scaleres function return's "false" ($?=1); i'm added "true" at end of scaleres() - and it's working
__
thank's for useful scripts
awaiting next release
CaptnBlack2008-01-30 02:59:46
Using the windows exe "Build Dec 17 2007, 17:32:16" I can't get stdin to work with Foobar2000s Converter or any other program that works with stdin.
Whoopie2008-01-28 11:20:57
Hi, I get some errors with the mks60 script:
Detecting crop area...
/usr/bin/mp4tools: line 75: "320" < "320" || "208" < "240" : syntax error: operand expected (error token is ""320" < "320" || "208" < "240" ")
Scaling a 320x240 movie to 320x240
/usr/bin/mp4tools: line 93: "320" > "320" || "240" > "240" : syntax error: operand expected (error token is ""320" > "320" || "240" > "240" ")
Best regards,
Whoopie
teknoraver2008-01-11 20:43:15
No, you can't force them.
I read them from the WAV header (first 44 bytes)
Can you send me the WAV header via mail?
Philippe2008-01-11 14:06:49
It seems not to be a Vista problem : same values on XP with the same file.
Do I forgot something ?
Philippe2008-01-11 14:01:03
Is there a way to force ch and sr values on Windows version?
teknoraver2008-01-09 18:03:05
indeed, they are very strange values,
but i never tried Vista before.
Philippe2008-01-09 18:00:08
On Windows Vista I use aacplusenc.exe and the answer is always :
No valid SBR configuration found for :
br=32000
ch=20294
sr=2797732
The value of br depends on the bitrate parameter, but for ch and sr, the values are always the same : 20294 seems to be a strange value !!!
Thanks for your help
Simon2008-01-08 22:06:48
Ok, It seems to work pretty well, but, I still have an issue :
If I try to encode with bitrate higher than 51kbps, I got this error (here with 52) :
No valid SBR configuration found for:
br=52000
ch=2
sr=22050
The file is 44100 with 2 channels...
Another issue:
You said earlier that you if bitrate is < 52, PS is used or, vlc pretend that it is used for bitrate < 45...
teknoraver2008-01-08 21:58:07
I added instructions on how to disable FFTW3
simon2008-01-08 21:49:18
Shame on me!!! I changed the makefile for testing and forgot to suppress the changes!
But I'm still searching for information...
Thanks
Simon2008-01-08 21:45:33
Hi there, I'm higly interressted by this encoder, more especially in his library. But, I have some trouble compiling it under mac os 10.4.11.
I had no problem with fftw3 and faac. But, when I run "make" for aacplusenc, I got this error :
cc -Wall -pedantic -O3 -ftree-vectorize -I/myComp/aacplusenc-0.12/libaacenc -I/myComp/aacplusenc-0.12/libbitbuf -I/myComp/aacplusenc-0.12/libfr -I/myComp/aacplusenc-0.12/libsbrenc -I/myComp/aacplusenc-0.12/libresamp -o aacplusenc aacplusenc.c -lm -L/myComp/aacplusenc-0.12/libaacenc -L/myComp/aacplusenc-0.12/libbitbuf -L/myComp/aacplusenc-0.12/libfr -L/myComp/aacplusenc-0.12/libsbrenc -L/myComp/aacplusenc-0.12/libresamp -laacenc -lbitbuf -lfr -lsbrenc -lresamp -lfftw3
/usr/libexec/gcc/i686-apple-darwin8/4.0.1/ld: Undefined symbols:
_fftwf_execute_dft
_fftwf_plan_dft_1d
collect2: ld returned 1 exit status
make: *** [aacplusenc] Error 1
Could you explain it? I have try the single and double precision version of fftw with no effect...
By the way, is there any information on the use off the library?
Thanks
teknoraver2008-01-08 21:42:48
mono encoding is limited to 44kbit, and is currently broken.
I'll fix it in 0.13
roy2008-01-08 17:42:36
How can i set the bitrate of aacplusenc.exe windows binary version?
It only works with an bitrate of 32.
If i try 48 it tells me:
No valid SBR configuration found for:
br=48000
ch=1
sr=22050
How can do other settings?
sharp2008-01-08 04:47:20
works for vodafone 904SH and 903SH.
Chris2008-01-03 05:01:21
thanks for reply, I have the following:
CPU: Intel(R) Core(TM)2 CPU T5500 @ 1.66GHz (Family: 6, Model: 15, Stepping: 6)
I definitely get a speed improvement on my system, I'll just edit the script locally.
thanks for the info
teknoraver2008-01-01 21:22:18
"threads=auto" gives slighty worse quality and NO speed increase, this on my Dual CPU system.
Also, for some obscure reason movies encoded so won't play on PSP
Chris2008-01-01 20:28:29
Thanks! I've been having problems getting anything encoded with faac to play on my ipod or in quicktime 7.3, this works.
you might add :threads=auto to the end of your x264opts to take advantage of dual core processors
teknoraver2007-12-29 15:48:27
Algorithm is in libaacenc and libsbrenc
Alx2007-12-29 11:40:44
Ok! thanks! I want to try to update algorithms. which files in the source should be replaced with updated to update aac encoding algorithms, sbr, ps, and other, affect the quality of coding?
teknoraver2007-12-28 22:11:59
You can compile under windows with cygwin, but I already have the win32 build so...
ANd, the algorithm updates very very rarely
And(2) low bitrates without PS will hear badly, so it's enabled by default
Alx2007-12-28 19:40:36
sorry that ask so many questions.. but how do I compile your source code under windows?
Alx2007-12-28 19:30:51
if i want to encode in down bitrate without ps? what can i do?
Alx2007-12-28 19:26:49
And in the releases, which are published in the [http://www.3gpp.org/ftp/Specs/html-info/26410.htm] coding algorithm updates or remain the same?
teknoraver2007-12-28 19:24:50
use bitrates >= 52kbit to don't use PS
Alx2007-12-28 19:19:13
I used stereo. I thought that in the m4a container, quality of sound a bit better. What is this ADTS format? there is a suggestion to make the optional argument #5 - 'nops' to be force disable ps. (may want) =)
teknoraver2007-12-28 17:55:09
you may be wrong, the encoding algorithm is still the same, just the IO routine and FFT are changed.
If you talk about mono encodings yes, they are broken
Alx2007-12-28 12:19:15
Thanks!
...
I found that the old encoder (v0.2) encodes better than the current one. Why?
teknoraver2007-12-27 18:55:09
you can use stdin and stdout using - as filename
Alx2007-12-27 18:51:37
sorry, i realized! ;\
and 'aacplusenc' can read wav data from stdin?
Alx2007-12-27 17:23:40
why compile 'aacplusenc' without fftw3 support? :(
Alx2007-12-27 14:25:21
Hi! how to that output file of 'aacplusenc' was in a raw aac format? what needs to be changed in the source code that was it?
teknoraver2007-12-25 02:55:33
mp4tools and nokiatagger are GPL
aacplusenc and amrenc are copyrighted by 3gpp
Smarter2007-12-24 19:02:46
What's the license of this project?
teknoraver2007-12-23 17:08:56
you have to use latest firmware from Sony
psp2007-12-23 16:11:09
dont works for psp :(
nix2007-12-14 10:07:30
thanks for this wonderful software.. I hope this works better than faac encoder for HE-AAC. love ya.. nix.
teknoraver2007-11-26 00:52:34
No there isn't.
SBR and PS part are encoded as ancillary data, so the only way to recon it is tryng to decode as AAC+ first,
then if it fails it's plain AAC
This is because usually AAC is wrapped in MP4.
Many muxers, most notably MP4Box, has a switch to mark the file as (HE-)AAC+
BTW, for technical questions contact me via mail
roman2007-11-26 00:42:36
and one more question :)
do you happen to know any good aac+ stream specification?
i need to know if aac+ file has SBR and PS to correctly report the samplerate and number of channels to a streaming server for accurate info.. ADTS headers don't contain the information about SBR and PS so i end up getting 22050hz/1 channel for files that are in fact 44100hz/2 channels (aac+v2)..
teknoraver2007-11-26 00:08:40
Never tried 32 bit, but it isn't supposed to work as the routine stores samples as 16 bit values
roman2007-11-26 00:03:29
just compiled, and successfully encoded a wav!
all went fine, thanks.
does it work only with 16bit wav?
i've also tried to encode a 32bit wav and this produced the file that is not audible...
Rohan Dhruva2007-11-18 20:18:07
Works great with my Nokia 6233. Thanks for this wonderful software and scripts ! Keep up the great work..
Dimitri2007-11-14 22:34:46
Works for nokia 6630 too
john2007-11-13 23:41:16
worked with a nokia 6230i
thanks
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