I have a few questions & ideas - potentially stupid, but they've been
bugging me. I'd try all the ideas myself except I can't get Lame to compile
& I don't have a clue how to implement them anyway.

1- Is it possible to change the sample rate by encoding frames using other
than 1152 samples? As an example, if we used a 44.1kHz WAV, making Lame
encode 2304 samples for each frame, purging all frequencies over 11025Hz or
the specified LPF, could we write it as a valid 22050Hz MP3 without
actually doing any resampling? (or something along those lines...) There'd
probably have to be a slight time/pitch shift for non-integer resampling
ratios, which would be worse for upsampling, but I think it might *sound*
better than resampling per sé. Normal resampling for upsampling, this
routine for downsampling?

2- Are some people saying Layer2 is actually better than Layer3 at the same
bitrates for some types of music? I wonder if quality could be improved by
switching layers midstream... Do MPEG standards support that?

3- Bit reservoir & Joint Stereo. Maybe this is already done, but just in
case it isn't... If switching between M/S & L/R modes lowers the quality,
then why not make the switch only when the new mode (not using the bit
reservoir) will be of better quality than the previous mode (using the
reservoir)?

4- In M/S encoding, approximately how much bandwidth is offered to the mid
channel & the side channel when M & S are of similar amplitude?

5- I think Lame would benefit if it could be forced to use short blocks
more readily when there's sharp attacks mixed with analog silence,
especially for lower sample rates. I have a sample where a lot of pre-echo
is introduced. I'm using 320kbit/sec for 44.1kHz, & 160kBit/sec for
22.05kHz, & the pre-echo is noticeable on both, especially the 22.05kHz
version... I think it's because the encoder isn't switching to short
blocks. And I'm sure it's *not* because of resampling - I LPFed both the
samples at 10kHz & they both sounded (almost) the same as WAVs.

6- What's the difference between normal stereo & dual channel apart from
normal stereo allowing a more "free" allocation of bandwidth between the
channels? In which circumstances would it be preferred over normal stereo?

Shawn
--
MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )

Reply via email to