[directmusic] Re: MP3 encoded wav tracks, a pain in the neck..
- From: "Scott Morgan" <scott@xxxxxxxxxxxxxxxxx>
- To: <directmusic@xxxxxxxxxxxxx>
- Date: Mon, 6 Jan 2003 20:49:39 -0600
I don't think I've ever heard the problem with the gap in audio 2 secs in.
I've encoded a lot of MP3 files. I have noticed that different apps and
sometimes different versions of the same app have unique behaviors in how
they encode stuff. Maybe there is a problem with the encoder app you are
using?
I did test the mp3 functionality in DX9 before I left MS and I remember it
being Ok..just with the annoyance of having to set the decompressed start on
the samples.
-Scott Morgan
http://Morganstudios.com
----- Original Message -----
From: "Jason Booth" <jason@xxxxxxxxxxxxxxxx>
To: <directmusic@xxxxxxxxxxxxx>
Sent: Monday, January 06, 2003 1:01 PM
Subject: [directmusic] MP3 encoded wav tracks, a pain in the neck..
>
>
> So my experience with MP3 encoded wav tracks continues:
>
> - I've verified on 3 computers (one running 2k, one running 2kserver,
> and one running XP) that the compression codec used in MP3 encoded wavs
> has some huge and obvious errors. The main one being a small but very
> audable gap in the audio placed about 2 seconds into every file.=20
>
> Also, the encoder doesn't seem to be included natively on winXP
> machines, which is extremely annoying considering this is the only way
> to deliver MP3 based playback with Direct Music. (note that the decoder
> is present).
>
> This combined with the offset errors makes working with mp3
> encoded wav tracks within direct music a real pain in the arse. My
> workaround is as follows:
>
>
> - Save the original track, record it's play time and sample count
>
> - Add a 2.1 second sine wav to the beginning of the track
>
> - compress the file as a wav-encoded MP3 using a third party program
>
> - Load the file into Direct Music, record it's play time and sample
> count
>
> - Subtract the difference between the new sample count and the original
> files sample count to find the offset
>
> - quickly realize that the max offset is about 37000 samples, far less
> than 2 seconds
>
> - Subtrack the time difference and use this to offset the wav file in
> the segment
>
> - visually check the file to make sure that the sine wav isn't present
> in the segment file, and hope the offset is accurate to not cause drift
> over time..
>
>
>
> I have a feeling that this still won't be accurate, simply
> because the time scale (in seconds.000) isn't going to be sample
> accurate. Somehow, I'll need to convert the time scale offset into
> sample counts, then see what the difference between the resulting sample
> count and my correct offset is, and use both offsets (time + samples) in
> direct music to get the correct offset.. Also, Direct Music reports the
> mp3 encoded wav as being a slightly different number of samples than
> Sound Forge does, which makes me wonder which count is correct.
>
>
> Now I have to ask the really annoying end user question: Did anyone
> actually try to use MP3 encoded wavs for anything even close to a real
> project before they shipped Direct X 9? I can't believe how convoluded
> this process is, or that anyone used MP3 encoded wavs without noticing
> the gap it places in every file 2 seconds into the stream.
>
>
>
>
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