[Mp4-tech] Re: Problem in compiling MPEG4 AAC Encoder and decoder..
help needed
Ralph Sperschneider
sps iis.fhg.de
Fri Sep 12 16:25:04 EDT 2003
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 While it is somewhat outdated and has not yet the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, number 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 package 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 <some-path>/audio/natural/refSoft/mp4mcEnc enc
make ISOMP4=0 -C <some-path>/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 <some-path>/data;
InfoAudio provides the following information:
WAVE file: <some-path>/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 <some-path>/data/test.wav -F noheader -D integer24
<some-path>/data/test.rawpcm
This gave the following output:
WAVE file: <some-path>/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: <some-path>/data/test.rawpcm
Sampling frequency: 44100 Hz
Number of channels: 1 (24-bit integer)
-read <some-path>/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 sampling 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
<some-path>/audio/natural/refSoft/bin/mp4mcEnc/Linux_g++_O0d/enc
<some-path>/data/test.rawpcm -r 64000 -c "-lc" -o
<some-path>/data/test_rawpcm.rawaac -d 3 -da 3
- ran the decoder
<some-path>/audio/natural/refSoft/bin/mp4mcDec/Linux_AFsp_g++_O0d/aacdec_mc
-s 44100 -Dn -p LC <some-path>/data/test_rawpcm.rawaac
<some-path>/data/test_rawpcm_rawaac.wav
- Checked <some-path>/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 options)
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 <sps iis.fhg.de>/* 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. 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/
More information about the Mp4-tech
mailing list