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Support - (fb2k) / Re: Foobar can't transcode OGG to MP3
Last post by crystal-clear -
Source audio codec is irrelevant but Vorbis is probably the only source format you tried that produces floating point output. Is it possible you didn't install the Free Encoder Pack to the same location where your foobar2000 v2.1.5 is? Or have you perhaps made foobar2000 use custom encoder paths? Only explanation I can think of is that you are actually using some decades old lame.exe that didn't yet support floating point intput.

Check that there are no wrong paths at Preferences -> Advanced -> Tools -> Converter -> Additional command-line encoder paths. By default it should be blank.

Thank you that's exactly it!! Somehow my foobar points to a codecs folder and the lame.exe inside was from 2010. Phew, finally!
Audio Hardware / Re: All DAC use PDM, even for PCM?
Last post by ktf -
And I remember a Philips CD player I owned in the late 1990s which said it was a 1 bit DAC or I also had a Technics with MASH which was (as i found out later) a 4 bit DAC.
That isn't really the whole truth.

When specifying a DAC, there are two main performance metrics: bandwidth and signal-to-noise ratio. If you say you want a 16-bit DAC, that doesn't really specify those two metrics. I can only assume you want a DAC of which the signal-to-noise ratio is better than what is offered by a 16-bit digital signal. As for bandwidth, I can only assume you mean 0 to 20kHz. If you want to play music for dogs, a higher bandwidth would be needed, which impacts DAC design.

Now, what is meant by that 1-bit or 4-bit MASH design, is that the DAC consists of a first stage that outputs a 1-bit or 4-bit signal. But that is only the first stage, not the final product. The point is, the are different ways to get the performance you want. You can use a very expensive first stage that is able to output 16-bit directly (I believe these are called non-oversampling DACs or NOS DACs) or you can use a first stage that outputs a few bits and run that at a very high sampling rate. This process is called delta-sigma modulation and is often referred to as 'oversampling'.

If all you care about is performance (signal-to-noise ratio), the best chips available seem to use the latter method, those are often called multibit delta-sigma DACs. See for example this datasheet. That is a very high performing DAC chip, with a THD+N of -115dB, which uses a multibit delta-sigma modulator. That thing is able to run in single-bit mode (DSD) but then it's performance will be slightly worse.

In the end, it doesn't really matter what is inside the device, as long as it does its job well enough. These chips are more then capable of not being the weakest link in the chain: what is often forgotten is that performance of analog-to-digital converters (ADCs) is not able to reach the levels DACs are. And often these figures are well beyond the capabilities of human hearing anyway.
3rd Party Plugins - (fb2k) / Re: foo_vis_spectrum_analyzer
Last post by sveakul -
Strange:  there is a stream Nordic Chillout Radio ( that according to Foobar v.2.2 x64 Properties is being sent at a sample rate of 7350 Hz--I find that hard to believe unless Foobar is resampling it internally (I did not have that sample rate in my "resample only" list in my active DSP SoX resampler), because the stream plays normally, although does not send metadata.  The stream's "icy-sr" header does show "7350" though that could be bogus. 

Anyway, what this has to do with foo_vis_spectrum_analyzer is that no spectrum is shown with this stream, although the Peak Meter component is active.  Why would that be??

This is one of those things that the sending server will probably fix tomorrow, but what would cause the no-spectrum?