does the problem happen when using directsound as output?
Also can you give more specs on your system, OS and how the DAC is connected?
Most of these game formats are notes of some sort, but converting them to MIDI requires specialized code to monitor the emulated sound hardware and convert the information into notes. I have written an SPC to MIDI converter, that produces one SPC file per unique sample, for easier instrument separation, but it still requires retuning and manual intervention.
foo_jesus is the only guaranteed way to keep stats for a session without user intervention. Otherwise, you'd need to manually trigger a configuration save periodically.
Usually a bad USB cable will cause the device to disappear and reappear in windows periodically. You'll have audio and then sudden you won't when a transfer gets corrupted and Windows resets the device.
What kind of noise or distortion does a marginal USB cable make?
I don't think that any of the common audiophile descriptions fit.
Those audiophiles aren't subjectivists, Floyd Toole ans Sean Olive are subjectivists.
The audiophiles are more neo-subjectivists.
The thing is, opinions about differences in sound quality are not welcome unless they can be backed up with evidence in accordance with our rules (have you read our rules?). If you don't have acceptable evidence, then don't make them.
I can't tell much from my side. I get an `std::runtime_error` from my decoder, and I bail out as it is assumedly hosed.
None of that is very clear. All of this should be possible with a regular audio editor (Audacity, etc.), if the filters are fixed. I'm guessing this is some kind of audio compression and that the filters are not fixed.
More specifically, optimum results have been achieved by using a filter having a 12 decibel per octave slope from 0 to 15,000 cycles per second.That doesn't make sense unless it's a variable frequency filter. i.e. 12dB per octave high-pass at 10kHz means something. A high-pass or low-pass filter has a single cutoff frequency. A bandpass filter has a low & high cutoff, but a 0-15KHz "bandpass" filter is a low-pass filter. If it's variable, what controls the cut-off frequency?
In one specific embodiment, this filter is formed by a pair of tuned amplifier circuits each having a 6 decibel per octave slope within the frequency range of interest.Again, not enough information.
In this embodiment, the speech waveform is preferably combined with a high frequency noise masking signal of lower amplitude prior to processing.What frequency, what level, and what's the frequency band of this "high frequency noise"?
Another normalizeIt's kind of unusual to clip after normalizing. Well... Since clipping is distortion it's unusual to intentionally clip at all... Sometimes you normalize after processing to prevent clipping.
For many, audio is a hobby - not a profession. Hobbies involve emotions so we always will find subjective comments in these forums, like this one, which is a crappy opinion coming from a low IQ man that can't back it p with scientific evidence or A/B tests on audio professional and enthusiast random samples.