So, what is the actual difference of a "subband codec" and a "transform codec"? Does a transform codec store the signal in frequency domain? Does a subband codec not? Do they have different filter banks? Do subband codecs use Fourier transform instead of mDCT to obtain the frequency bands? And what actually makes MP3 a hybrid codec?
What is the essential difference between splitting a signal into subbands through a filterbank and retrieving frequency information through an MDCT? Both give information about the level of a signal in a certain frequency range (more or less) over a block of data.Also, what would be the point of applying an MDCT to a signal that is already split into subbands, and to what exactly is the MDCT applied? (each of the subbands, some strange mix or ...?)
If you're talking about MP3 specifically, I don't think there is actually any point at all applying mDCT for the subband data. Layer 3 had to be backwards-compatible with Layer 2, and thus the subband/transform hybrid structure. A bad decision from MPEG organization, say most people. As well as the missing scalefactor issue for the highest frequency band. Luckily we have AAC now
Not really true.. You must remember that at the time MPEG1 is standardized, computational power is still limited.. A subband based structure has its advantages in such as the possibility to implement a scaleable MP3 decoder even though the bitstream itself is not scaleable.
Of course the complexity of the decoder will be halved if you choose a 22050 Hz sampling rate decoding.. The rest of the modules are halved in complexity...
QuoteNot really true.. You must remember that at the time MPEG1 is standardized, computational power is still limited.. A subband based structure has its advantages in such as the possibility to implement a scaleable MP3 decoder even though the bitstream itself is not scaleable.How? Is it just by only applying the inverse-qmf on only half of the subbands?Would this really be significant, regarding processing power?
You have not seen a scaleable MPEG1 Layer II decoder? I have seen it back in 1996.. In those days, silicon is expensive! But today, it would be a different story..
Of course, the dequantization module involves a power function. If you are going to implement it in fixed point DSP, this power function is going to be very complex unlike the floating point enhanced performance of the Intel Processor...
I remembered something.. The MP3 specs allows the block switching to occur above certain subbands.. It is something like mixing short blocks with long blocks unlike MPEG4 AAC SSR which you MUST switch all 4 subbands simultaneously!Well, this might be useful, considering that you can detect the attacks from the time domain subband output samples and decide which subband should be switched to short block!
No.. I was talking about scaleable MPEG1 Layer III decoders - and so far I haven't heard about them? Maybe you can point me to some vendors making that stuff?
True, but IIRC - mixed blocks in MP3 had some other nasty limitations that prevented them to be used in default encoding modes of many encoders.