Support Head-Fi.org by
starting all of your
Amazon.com shopping by
clicking here.
____________________________________________________________________
Today's Featured Head-Fi Blog: Jude's Blog
____________________________________________________________________
Please help
support Head-Fi by becoming a Contributing Member
CLICK
HERE -- Contributing Members, thank you
for your generous support! --
In Winamp I use an ASIO output plugin that has SSRC built into it. It works fine there, but ofcourse...no linear mode. The winamp version doesn't have a hardware mixing option either, though.
__________________
They gave me his photo, threw me in a pizza oven, called it a "pod", and told me to wing it
Whee. So well... Does the F2000 SSRC do linear resampling?
The current versions of SSRC lacking an option for software or hardware resampling might mean it automatically switches if hardware resampling fails, or just does everything in software (it is safer, and hey, CPUs nowadays are powerful...).
According to Wiki, http://en.wikipedia.org/wiki/Free_Lossless_Audio_Codec , FLAC supports non-floating-point bit depth of up to 32-bit and any sampling rate up to 1048570 Hz. So as long as the wave format is integer-based, an external editor should be capable of linear resampling, then FLAC could re-encode the 192-KHz waves.
1. A stack of Winamp DSP adapter for F2000+Adaptx 3.5 DX adapter+some kind of Directx linear resampler.
2. The same stack, except running natively in Winamp. DSP processing is done before output. Hence whatever the DSP resampler's going to pass to the output plugin, won't be affected by the plugin's resampling.
3. A linear resample in an external wave editor, then re-encoding to FLAC.
Hancoque, what artifacts are you talking about, and what upsampling rate did you use?
I created 32 bit float samples at 44.1 kHz and upsampled these to 192 kHz using the different plugins and also Adobe Audition. All calculations took place in 32 bit floating point precision (including foobar2000).
The wave view for the original and upsampled waves is not accurate. The software you're using is using interpolation between samples when it's displaying the waveform, automatically. The wave view for "original" should look very much like the one for SRC linear.
The spectral analysis seems accurate though. As expected, the linear upsampling mode measures poorly.
__________________
They gave me his photo, threw me in a pizza oven, called it a "pod", and told me to wing it
The wave view for the original and upsampled waves is not accurate. The software you're using is using interpolation between samples when it's displaying the waveform, automatically.
Yes, but the original file is a perfect sine wave when played back, the one upsampled by Adobe Audition is too but the one upsampled linearly isn't. The program shows you what you actually hear. Linear connections between each sample don't represent the real world.
Originally Posted by b0dhi
The spectral analysis seems accurate though. As expected, the linear upsampling mode measures poorly.
The artifacts you see in the spectral view are the linearly interpolated points. They add audible distortions. Why would you want to have these?
If so, no, it doesn't.
Linear Mode
