hello, every body
Iam new to audio processing and iwant to do research on aac . Iam downloaded Facc 1.64 encoder and debugging w.rt std 13387-7 . now I wnt to seperate Tns and filterbank in the code to test . my doubt is can i use c or matlab . if i use matlab how can I start , and aac filterbank is cosine modulated filterbank is polyphase or qmf type. please reply me these ansewrs

Thank U

Rangababu
Hyderabad
India

my doubt is can i use c or matlab . ?
>> Yes you can use C or Matlab. ISO AAC Encoder can also serve as good starting point for AAC.
if i use matlab how can I start ?
>> to start in Matlab you need to do following
- read 1024 or 960 points from a wav
- do a frequency transform (MDCT) on input samples. Make sure you overlap 50% of current and preceeding sample before MDCT. MDCT is similar to FFT in implementation respect but a mere pre and post twiddle calculation. Look in the aac reference code for pre post twiddle operations.
- Now calculate a quantization factor for each band using
quantFactor(sfb) = pow(2.0, 0.1875*(scalefactor(sfb) - commonSF)); /*let scalefactor(sfb) =0 in the beggining and let commonSF=be big enough to quantize largest dct coefficient to 8K*/
- Since you dont need to do synthesis as a new bie, simply quantize each line as follows
quantizedSpectrum(w)=(Spectrum(w)^(3/4)*quantFactor + MAGIC_NUMBER);

After this point starts the noiseless coding and preparation for bit formatting so if you can implement your model till here in matlab, you will do good. Although bit allocation is critical to any encoder but thats secondary for you now, you have to do the bit allocation(e.g. rate/dist loop) to finish your encoder anyway.


Hope this helps,

Regards,
Nabsha