[Mp4-tech] Re: MPEG4 AAC .. help needed.
Ralph Sperschneider
sps iis.fhg.de
Mon Sep 15 20:32:34 EDT 2003
sumit thatte wrote:
> Dear Sir,
> I am very much thankful to you for your kind help. I tried all the
> guidelines given by you for running the ".pcm" (or Raw) file. But it is
> giving the same problem. I am using WINDOWS 2k platform. Pls help me sir...
It is most likely not a problem of the operating system. I personally prefer the
Unix environment, so if I have only Windows available I am running the Cygwin
environment.
>
> (a)Will it be a problem with the test vectors ? I downloaded it from the
> ISO site.
> http://www.iso.ch/iso/en/ittf/PubliclyAvailableStandards/ISO_IEC_14496-4_2000_Conformance_Testing/CONFORMANCE_BITSTREAMS_CD2/AUDIO/GA/AAC_LTP/
It is most likely not a problem of the test sequences.
>
> I wanted to zip the test vector and send it to you but it is large so i
> didn't do it
I appreciate that you didn´t send it by e-mail. Downloading is easy for me, but
huge attachment overcrowd my mail folders.
> , but can you pls down load it and test it ?. Testvector
> Name - AP01_48.pcm
I did this.
> Pls tell me sir do you observe the clipping when you give to decoder.
I observe clipping, if I treat the raw pcm data in the wrong way, otherwise I
observe a clear sine sweep.
>
> (b) I set the MP4_RAWAUDIOFILE environment variable to "integer16, 0,
> 48000.0, native, 2, 1.0" It gives the Clipping error or huff decode error..
That might be.
> (c)I set the MP4_RAWAUDIOFILE environment variable to "integer16, 0,
> 48000.0, native, 1, 1.0" It gives the Clipping error or huff decode error..
That might even be.
>
> (d) I tried with "integer24, 0, 44100.0, native, 1, 1/256" but it gives
> the same error and also says It says "AFset - Fixed up, non integer
> number of samples"
That again might be.
BTW: The correct string for this file would be:
"integer16, 0, 48000.0, big-endian, 1, 1"
So you e.g. could convert this file into a wav-file using the following cmdline:
CopyAudio -t noheader -P "integer16, 0, 48000.0, big-endian, 1, 1.0" AP01_48.pcm
ap01_48.wav and listen to it using e.g. the MediaPlayer
Note: CopyAudio comes with the AFsp library package, so you should have it as
well as the appropriate documentation to find the proper settings on your own
(the string required by the encoder is excactly the string required by the AFsp
library ;-) to open the raw pcm waveform with the correct parameters).
>
> (e) I tried giving the test vectors from the site mentioned by you.
> (ftp://mpaudconf:[email protected]/mpeg4audio-conformance)
> I have used ap01_48.mp4 test vector for decoder.
> Still it gives error "Huff decode".
In order to decode the file ap01_48.mp4 you need two MPEG4 specific
prerequisites: LTP support and mp4ff support. Both prerequisites are not
supported by the executable aacdec_mc (this supports just MPEG-2 AAC, decoding
raw, adif and adts formats with Main, LC and SSR profile), but only by mp4audec_mc.
You will need the isomediafile library to successfully build the executable
mp4audec_mc. To avoid further questions on this:
- download and unzip
http://www.iso.ch/iso/en/ittf/PubliclyAvailableStandards/ISO_IEC_14496-5_2001_Software_Reference/systems.zip
- compile the isomediafile library: make -C
<some-path>/systems/mp4_file_format/libisomediafile/<platform>/libisomediafile
- set the environment variable ISOMP4_PATH, on a tcsh it looks as follows:
setenv ISOMP4_PATH <some-path>/systems/mp4_file_format/libisomediafile
- build the decoder: make -C <some-path>/audio/natural/refSoft/mp4mcDec
ISOMP4_NAME=isomediafile mp4audec_mc
- to encode LTP, you need to compile the encoder as follows:
make -C <some-path>/audio/natural/refSoft/mp4mcEnc ISOMP4=0 DEFFLAGS=-DMPEG4V1
>
> I have a couple of test vector from SQAM (Software quality accessment
> material) site in .wav format. The code is encoding and decoding them
> properly.
This is great news.
>
> Sir,
> My questions are
>
> (1) Are these ISO testvectors for LC and LTP are right ?
The new ones are right, the old ones are outdated (but still sufficient for
tests like yours).
> (2) I want to implement this code on 16 bit FIXED point processor , so
> which of the settings are correct ?
> [ i ] "integer16, 0, 48000.0, native, 2, 1.0" *OR*
> * *[ ii ] "integer24, 0, 44100.0, native, 1, 1/256"
These settings are not directly related to your planned implementation but
rather to the available raw pcm data. The old raw pcm reference waveforms have
an accuracy of 16 bit, the new waveforms (in wav/riff format) have an accuracy
of 24 bit. In my previous example I used a 24 bit WAV waveform and converted it
inte a 24 bit RAW waveform. Thus I was using "interger24" at that time, but for
16 bit raw files you have to use "integer16". At the same time I used 1/256 as
scalefactor, while for 16 bit raw fiels you have to use 1 (or with the AFsp v8,
you can simply set the scalefactor to "default" and should not have to take care
about the proper setting here on your own; check out the afsp documentation for
details). You of course have to set the sampling frequency appropriate (which
for the aac conformance test sequences is given in the file name).
>
> Can you please guide us why i am not able to run (.pcm/ .raw) files ?
Having the executables ready and the environment variable MP4_RAWAUDIOFILE set
correctly, the following lines should work fine (remember to re-build the
encoder as outlined above):
<some-path>/audio/natural/refSoft/bin/mp4mcEnc/Linux_g++_O0d/enc
<some-path>/data/AP01_48.pcm -r 64000 -c "-ltp" -o
<some-path>/data/ap01_48_pcm.rawaac
<some-path>/audio/natural/refSoft/bin/mp4mcDec/Linux_AFsp_isomp4_g++_O0d/mp4audec_mc
-s 48000 -Dn -p LTP <some-path>/data/ap01_48_pcm.rawaac
<some-path>/data/ap01_48_pcm_rawaac.wav
>
> Pls can you help me on this sir.. I need your help desparately.
I tried to do this and already spend a significant amount of time on this
(Working with a 2 year old software is not always fun, too).
BTW: We (FhG) offer AAC encoder and decoder implementations for fixed point
DSPs. Please contact Mr. Stefan Geyersberger for details.
Best regards,
Ralph
>
> Waiting for your reply.
>
> Thanks and regs
> Sumit Thatte
>
>
>
>
> */Ralph Sperschneider <sps iis.fhg.de>/* wrote:
>
> Dear Sumit,
>
> sumit thatte wrote:
> > Dear Sir,
> > First of all I am very much thankful to you for your kind and timely
> > help. It will be very much favourable for me if you pls further
> help me
> > tackling the problem. Pls help sir.
> >
> > The secnario is like this,
> > Our objective is to do the AAC codec for *LC* and *LTP* profile only.
> > I have downloaded MPEG4 AAC codec code from ISO site link given
> below,
> >
> >
> http://www.iso.ch/iso/en/ittf/PubliclyAvailableStandards/ISO_IEC_14496-5_2001_Software_Reference/audio.zip
> Okay, thats the 2001 edition. While it is somewhat outdated and has
> not yet the
> codec_doc subdirectory, it should nevertheless serve your needs.
>
> > I extracted it and used the code in the folder
> > \audio\natural\refSoft\mp4mcEnc\src for encoder and
> > audio\natural\refSoft\mp4mcDec\src for decoder. I have attached
> both the
> > zip files with this mail. I have sent workspaces also.
> >
> > The compile time options we can see in projects.. settings.. C/C++
> > and run time in projects.. settings.. Debug.
> It turns out that you are working with Windows. However, Windows is
> natively not
> suported by the MPEG-4 Natural Audio reference software, but should
> nevertheless
> work seemlessly (assuming that the necessary ds[pw] files are
> prepared). I will
> try to retrace you problem under Linux.
>
> > I am able to compile the project and able to encode and decode for
> > *.wav* files correctly
> This is great news.
>
> > but I am not able to encode and decode ".pcm" files.
> Well, I am not sure what you mean with ".pcm" files. Wave-Files also
> contain pcm
> data, but additionally provide a certain header storing the
> appropriate meta
> data (sampling frequency, number of channels, nu! mber of bits per
> sample, file
> length, little endian or big endian, ...). I however assume that you
> mean files
> containing raw pcm data by refering to ".pcm" files. In that case
> you of course
> have to provide that missing meta data explicitly to the encoder, in
> that case
> by means of an environment variable.
>
> > *For .pcm file, if the encoder encodes with LC option and
> generate .aac
> > file when i supply it to decoder having LC or LTP it says
> "CLIPPING the
> > data" or gives the error "Huffdecode".*
> > **
> > The relevent test bit streams i have downloaded from the
> following link
> > for LC and LTP profile.
> >
> >
> http://www.iso.ch/iso/en/ittf/PubliclyAvailableStandards/ISO_IEC_14496-4_2000_Conformance_Testing/
>
> Okay, these are the still official test sequences, but you might
> want to
> download the up-to-date sequences from
>
> ftp://mpaudconf:[email protected]/mpeg4audio-conformance
>
> > I am using AFSP p! ackage version v6r8.
> I didn´t yet try the newest version of this lib (I am still running
> v6r7b), but
> the most recent version shouldn´t cause any problem.
>
> > My questions are ,
> >
> > (a) Am I using the right code for LC and LTP profile.?
> Yes.
>
> > (b) How to encode and decode .pcm bit streams?
> Here is what I did:
> - compiled enc and aacdec_mc by running
> make ISOMP4=0 -C /audio/natural/refSoft/mp4mcEnc enc
> make ISOMP4=0 -C /audio/natural/refSoft/mp4mcDec aacdec_mc
> (ISOMP4=0 just to avoid the necessity of the mp4 file library, which
> is not
> required in the current situation)
> - took a wave file test.wav into a subdirectory /data;
> InfoAudio provides the following information:
> WAVE file: /data/test.wav
> Number of samples : 361472 (8.197 s) 2023-09-13 14:37:12 UTC
> Sampling frequency: 44100 Hz
> Number of channels: 1 (24-bit integer)
> File name: test.wav
> Header length: 80
> Sampling frequency: 44100
> No. samples: 361472
> No. channels: 1
> Data type: integer24
> File byte order: little-endian
> Host byte order: little-endian
>
> - generated a raw pcm file as follows:
> CopyAudio /data/test.wav -F noheader -D integer24
> /data/test.rawpcm
> This gave the following output:
> WAVE file: /data/test.wav
> Number of samples : 361472 (8.197 s) 2023-09-13 14:37:12 UTC
> Sampling frequency: 44100 Hz
> Number of channels: 1 (24-bit integer)
>
> Headerless audio file: /data/test.rawpcm
> Sampling frequency: 44100 Hz
> Number of channels: 1 (24-bit integer)
>
> -read /audio/natural/refSoft/mp4mcEnc/readme/README:
> "
> In order to use raw data files as input the AFopenRead function
> requires that the MP4_RAWAUDIOFILE environment variable be set
> as follows:
>
> setenv MP4_RAWAUDIOFILE "integer16, 0, 48000.0, native, 1, 1.0"
>
> where the 48000.0 is the sampli! ng rate, the 1 means one channel,
> and the 1.0 means a gain of 1.0. Other sampling rates can be
> used, but I think that currently the psychoacoustic model only likes
> 44100 and 48000. For stereo operation, the MP4_RAWAUDIOFILE
> variable should be set as:
>
> setenv MP4_RAWAUDIOFILE "integer16, 0, 48000.0, native, 2, 1.0"
> "
>
> - since this explanation is not fully suficient, I read the
> appropriate AFsp
> documentation (AFsp-v6r8/html/libtsp/AF/AFsetNHpar.html). Here, the
> form of
> the list is specified as "Format, Start, Sfreq, Swapb, Nchan,
> ScaleF" and the
> individual parameters are explained in detail.
>
> - set the environment variable appropriately (in my tcsh) as
> setenv MP4_RAWAUDIOFILE "integer24, 0, 44100.0, native, 1, 1/256"
>
> - ran the encoder
> /audio/natural/refSoft/bin/mp4mcEnc/Linux_g++_O0d/enc
> /data/test.rawpcm -r 64000 -c "-lc" -o
> /data/test_rawpcm.rawaac -d 3 -da 3
>
> - ran the decoder
> /audio/natural/refSoft/bin/mp4mcDec/Linux_AFsp_g++_O0d/aacdec_mc
> -s 44100 -Dn -p LC /data/test_rawpcm.rawaac
> /data/test_rawpcm_rawaac.wav
>
> - Checked /data/test_rawpcm_rawaac.wav and found that it contains
> audible data similar to what I used as input pcm data. You of course
> can also
> decode towards a raw pcm file (although it is most times a bad idea
> to have
> audio data around with no information about their properties) by not
> using the
> .wav extension (which automatically enforces the decoder to write
> the WAV header).
>
> > (c) The test bit streams i am using are OK or not ?
> As I said above, newer ones are available at the fhg server.
> However, the test
> sequences are only good to test the decoder. To setup the codec
> chain you can
> use any audio data.
>
> > (d) What are the *necessary and sufficient* testing scenarios for
> LC and
> > LTP profile? (i mean run time option! s)
> The run time options are explained above. However, that has nothing
> to do with
> (conformance) testing. Please read ISO/IEC 14496-4 for details about
> testing.
>
> > (e) Are there any other bits streams available for LC and LTP
> profile?
> The test sequences above are official sequences specified and
> generated by MPEG
> (again, please use the updated sequences as available on the FhG ftp
> site).
> Beside this, you can encode any audio data with any parameters and
> by this
> create any number of AAC bit streams ;-)
>
> Best regards,
>
> Ralph
>
> >
> > Thanks for you timely help agin Pls help me regarding this quaries..
> >
> > I will be very thankful to u sir, eagerly waiting for youy kind
> reply.
> >
> > Thanks and regards
> > Sumit Thatte
> >
> >
> >
> > */Ralph Sperschneider /* wrote:
> >
> > Dear Sumit,
> >
> > sumit thatte wrote:
> >
> > > Hi all,
> > > ! I have downloaded the source code for MPEG4 AAC(advanced audio
> > codec)
> > > from ISO site.
> >
> > Where did you download it? To evaluate your problems I need to have
> > exactly the
> > software that you have.
> >
> > > I want to run it for LC and LTP profile. I am using AFSP
> > > package v6r8.
> > > I am not able to encode and decode ".raw" files.
> >
> > Saying this you admit that you are able to compile the code, right?
> > Which
> > encoder/decoder are you working with? The provision of some command
> > lines you
> > tried might allow me to shed some light into your problems.
> >
> > Ralph
> >
> > > Pls can somebody tell me what are the compilation, run time
> > options and
> > > the environment variables to set?
> > > Pls help me , I am badly stuck here.
> > > Thanks in advance.
> > > Regards
> > > Sumit Thatte
> --
> Dipl.-Ing. R! alph Sperschneider | Phone: +49 9131 776 344
> FhG IIS | Fax: +49 9131 776 398
> Am Wolfsmantel 33 | mailto:sps iis.fhg.de
> D 91058 Erlangen | http://www.iis.fhg.de/amm/
>
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--
Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344
FhG IIS | Fax: +49 9131 776 398
Am Wolfsmantel 33 | mailto:sps iis.fhg.de
D 91058 Erlangen | http://www.iis.fhg.de/amm/
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