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Re: AudioConverter MP3 decode questions
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Re: AudioConverter MP3 decode questions


  • Subject: Re: AudioConverter MP3 decode questions
  • From: William Stewart <email@hidden>
  • Date: Wed, 15 Mar 2006 12:37:27 -0800


On 14/03/2006, at 4:25 PM, Vince Shumaker wrote:

Thought I'd also forward this to the list...

Thanks!

How time a given packet represents is described by the ASBD - most
formats specify a frames per packet (for linear pcm this is 1, for
compressed its generally > 1). For instance, AAC typically represents
its data as 1024 frames/packet - apple lossless as 4096, etc...

...and for MP3 its 1152? I'd like to clarify what this should be when I create an AudioConverter using ASBD. I know that MP3 frames have a fixed number of 1152 samples/frame. I've also seen some of the /Developer examples using AudioFile api to get the ASBD... and the frames/packet value comes back as 1152.

So I guess that I shouldn't think about "samples per frame" vs "frames
per packet" when describing MP3. It's really just a blob of time
expressed as 1152?

Its a blob of data that represents a period of time -> frames per packet / sample rate (to give you that time in seconds), But yes, you got it.


There's some utility calls in the AudioFormat API to fill out ASBD - have a look at the utility code for the ConvertFile example in the SDK. By using the AFormat calls, you don't have to hard code this stuff in your own code.

Bill

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References: 
 >AudioConverter MP3 decode questions (From: "Vince Shumaker" <email@hidden>)
 >Re: AudioConverter MP3 decode questions (From: "Alexander v. Below" <email@hidden>)
 >Re: AudioConverter MP3 decode questions (From: philippe wicker <email@hidden>)
 >Re: AudioConverter MP3 decode questions (From: "Alexander v. Below" <email@hidden>)
 >Re: AudioConverter MP3 decode questions (From: William Stewart <email@hidden>)
 >Re: AudioConverter MP3 decode questions (From: William Stewart <email@hidden>)
 >Re: AudioConverter MP3 decode questions (From: "Vince Shumaker" <email@hidden>)

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