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enhanced aac+ to aac lc
wind
post Nov 2 2012, 09:53
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Hi, everyone, has there anybody used enhanced aac+ code(TS 26.410) to encode/decode only aac lc ,without sbr?

I modified the encoder of the enhanced aac+, ,there is no sbr, then when the decoder decode the new .3gp file ,no sbr is decoded, but the outout sample rate is 1/2 downsampled, because aac lc works at 1/2 sample rate, the sbr works at original sample rate, and in the decoder ,it is sbr make it become to original sample rate, but now there is no sbr.
Does anyone know how to make the sample rate of decoder output same as the original input file sample rate, i tried to modify the encoder to encode at original sample rate,but there will be no sound, and if i use 1/2 sample rate .3gp & modify the decoder to change the sample rate and frame size, the sound will sound strange...
or
is that ok if i upsample the new decoder output file (1/2 sample rate ) through matlab?

Thanks in advance.

This post has been edited by wind: Nov 2 2012, 10:29
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Dynamic
post Nov 3 2012, 15:43
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I guess we need to inspect the source code of an open source decoder to find out. I can't see how mathematically they could generate with an iMDCT containing twice as many samples as 'frequency' bins (positive and negative 'frequencies' both count towards the total) provided in the transform domain. Therefore, I'd assume it must be done after converting to the time domain (i.e. by upsampling with appropriate anti-alias filtering).

I'd like to be enlightened by anyone who knows for sure.
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wind
post Nov 4 2012, 12:13
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QUOTE (Dynamic @ Nov 3 2012, 15:43) *
I guess we need to inspect the source code of an open source decoder to find out. I can't see how mathematically they could generate with an iMDCT containing twice as many samples as 'frequency' bins (positive and negative 'frequencies' both count towards the total) provided in the transform domain. Therefore, I'd assume it must be done after converting to the time domain (i.e. by upsampling with appropriate anti-alias filtering).

I'd like to be enlightened by anyone who knows for sure.


Hope someone can read this post.

I don't know how to write upsampling with filtering in C code, so maybe i can only use some software to convert sample rate.
About the sample rate conversion software, how about the Adobe Audition software?which one do you think is better?
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