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[sibling]

I bought MJ 8 a few months back. When I try to convert from WAV format to MP3 (or even to OGG), the resulting file plays at probably four times normal speed. Strangely enough, I can convert WAV to WMA (Windows Media Player format) and it works just fine. How can I make a good conversion from WAV to MP3? This was actually one of the two main reasons I bought the program, and now I can't get it to work right.

[KingSparta]

I think It Is The Sample Rate That Does That No Matter What Format It Is In.

[sibling]

Sample rate made no difference. MP3 and OGG always play extremely fast (high-pitched, like The Chipmunks on meth), but WMA plays fine. Please help me resolve this, as it's one of the main reasons I bought MJ8.

[Alex B]

So what has changed on your PC? I guess you tried MJ8 before bying it. I didn't have that kind of problem when I used MJ8.

Is it only files that are converted from wave format? Do you have any other MP3 or Ogg Vorbis files that play normally? What is the source of the wave files? Are they ripped from Audio CDs at 16-bit/44.100 kHz?

Previously I have noticed that the WMA encoder automatically resamples files of a different sample rate to 44.100 kHz unless the encoding program provides a separate option for setting a different sample rate. (MJ and many other programs don't have that option.) The MP3 and Vorbis encoders don't automatically resample. This may be connected with the behavior you have experienced.

Can you provide a small sample file?

Please post your system info from the MJ help menu too.

[sibling]

So what has changed on your PC? I guess you tried MJ8 before bying it. I didn't have that kind of problem when I used MJ8.

Yes, but I never tried file conversion. I just assumed it would work. I tried it for the first time earlier this week and got bad results.

Is it only files that are converted from wave format?

No. For example, if I try to convert an MP3 file to OGG, same thing happens.

Do you have any other MP3 or Ogg Vorbis files that play normally?

All of my original files play normally, in whatever format. It's the conversion to MP3 (or OGG, which I'm not actually interested in) that makes the files unlistenable.

What is the source of the wave files? Are they ripped from Audio CDs at 16-bit/44.100 kHz?

Yes.

Previously I have noticed that the WMA encoder automatically resamples files of a different sample rate to 44.100 kHz unless the encoding program provides a separate option for setting a different sample rate. (MJ and many other programs don't have that option.) The MP3 and Vorbis encoders don't automatically resample. This may be connected with the behavior you have experienced.

Since I'm ripping WAV files at 44,100 stereo, I don't think this would apply.

Can you provide a small sample file?

I'd be happy to, if you can tell me how to attach files. Or I could email you some files.

Please post your system info from the MJ help menu too.

Media Jukebox PLUS 8.0.400
Install Path: C:\Program Files\J River\Media Jukebox\

CPU: AMD Athlon 1594 MHz MMX
Memory: Total - 523 MB, Free - 240 MB
OS: Microsoft Windows XP  Workstation 5.1 Service Pack 2 (Build 2600)

Internet Explorer: 6.0.2900.2180
ComCtl32.dll: 6.00.2900.2180
Shlwapi.dll: 6.00.2900.2753
Shell32.dll: 6.00.2900.2763
wnaspi32.dll: Internal ASPI Layer

[Alex B]

You could send a couple of sample files (e.g. mp3 and ogg) to:  alexb_at_luukku.com  (replace _at_ with @)

Please select small files.

I have no idea what causes your problem, but at least I can check the file formats.

[sibling]

Okay, I sent you the fiiles yesterday. Please tell me if you need me to resend them.

[Alex B]

The wave file is a 22.05 kHz mono file.

The converted mp3 file is a 44.1 kHz stereo file.
Here is what Encspot shows:

Bitrates:
----------------------------------------------------
128     ||||||||||||||||||||||||||||||||||||||||       98.0%
----------------------------------------------------

Type                : mpeg 1 layer III
Bitrate             : 138
Mode                : joint stereo
Frequency           : 44100 Hz
Frames              : 50
ID3v2 Size          : 1631
First Frame Pos     : 1631
Length              : 00:00:01.3
Max. Reservoir      : 228
Av. Reservoir       : 50
Emphasis            : none
Scalefac            : 3.5%
Bad Last Frame      : no
Encoder             : Lame 3.92

Lame Header:

Quality                : 55
Version String         : Lame 3.92
Tag Revision           : 0
VBR Method             : cbr
Lowpass Filter         : 15400
Psycho-acoustic Model  : gpsycho
Safe Joint Stereo      : no
nogap (continued)      : no
nogap (continuation)   : no
ATH Type               : 2
ABR Bitrate            : 128
Noise Shaping          : 1
Stereo Mode            : Joint Stereo
Unwise Settings Used   : no
Input Frequency        : 44.1kHz

Another diagnostic tool (Audio Indentifier) shows this LAME tag:

Tag revision:        0
Version string:      3.92
Quality:             55 (V4 and q5)
Encoding method:     cbr
Lowpass:             15.400Hz
RG track peak:       <not stored>
RG track gain:       <not stored>
RG album gain:       <not stored>
nspsytune:           no
nssafejoint:         no
nogap continued:     no
nogap continuation:  no
ATH type:            2
Bitrate:             128
Encoder delay:       576 samples
Padded at end:       1.771 samples
Noise shaping:       1
Stereo mode:         joint
Unwise settings:     no
Source sample freq:  44.1kHz
MP3Gain change:      <none>
Preset:              <not stored> - unable to guess
Surround info:       none
Music length:        20.897 bytes
Music CRC:           F9B9
Actual Music CRC:    F9B9
Info tag CRC:        7943
Actual Info Tag CRC: 7943

Code: [Select]

File Name Type Size Length Ratio Bitrate Encoder Info Sample Rate Channels Bits Per Sample
JtC.wav WAV 0,21 MB 00:05 100,00% 353 kbps Windows PCM 22050 1 16
JtC.mp3 MP3 0,02 MB 00:01 10,69% 128 kbps MPEG 1 Layer III 44100 2 16


For some reason the MP3 encoder has made a 44.1 kHz file without resampling the data. Something else seems to be wrong besides that because the duration is only 1/4 of the original.


Att: Matt  (from JRiver),

Could you check these files? Would a custom LAME command line help?

Here are the samples:  JtC.wav  &  JtC.mp3

[Alex B]

I found this old thread about the same issue: http://yabb.jriver.com/interact/index.php?topic=18300.0

Obviously you need to set a custom command line when converting 22.05 kHz mono files to MP3 or Ogg Vorbis formats. As I said before the WMA encoder resamples the files automatically to 44.1 kHz. That's the reason why WMA works.

I can't test this with MJ8, but for MP3 files you could try this custom command line:
-b 64 -m m -s 22.05    or this:  -b 64 -m m -s 22.05 --resample 22.05

I think 64 kbps is more than enough for this kind of speech file because the source file is mono and has a low sample rate. I quess even 32 kbps would sound pretty much the same.

The LAME command line reference guide: http://lame.sourceforge.net/USAGE

[DWB12K]

I am expericing the same problem with a High-pitched conversion.  Did anyone get the right fix here?
kingsley@member.afa.org