cesium wrote:I think that most of the ignorance out there is rational - after all, they have people like us to help, right? The crazy stupid people are us who bother with this for free, and therefor we are far too stupid to be running anything (I do recall Eisenhower also warned against a scientific-technological elite as well as the Military-Industrial complex in his exurgal speech).
It seems that you like to help. Perhaps, it is nice to be busy with this. However, a kind of LiveCD together with a comprehensible "howto" may essentially reduce the number of questions, and we may have enough time to discuss ideological problems.
Ideology has a practical value, simply because nothing is free. Linux may cost a lot of your valuable time. You may agree to spend your time for this, if the promise is freedom, namely, the freedom of action, that is, the freedom to do what you want and how you want. However, sooner or later, a Windows refugee may arrive to OSS4 forum, where he may learn an "unpleasant truth", which might be revealed in this way: "You will never get the much desired freedom, because you are not intelligent enough to remove PulseAudio".
cesium wrote:That should appear somewhere in the chip's sheet from intel.
Such intel sheets usually claim that you can get with Intel HDA anything what you can imagine.http://www.intel.com/design/chipsets/hdaudio.htm
Intel HD Audio delivers significant improvements over previous generation integrated audio and sound cards. Intel HD Audio hardware is capable of delivering the support and sound quality for up to eight channels at 192 kHz/32-bit quality, while the AC‘97 specification can only support six channels at 48 kHz/20-bit.
I certainly want to have 192 kHz/32-bit quality with ICH6, but what I actually have with Linux drivers for ICH6 is 48 kHz/32-bit quality. I also want to have HW mixing with ICH6. I do have it with ICH4 with ALSA and OSS4. I am not going to take your "market theory of soundcards" for granted. I am not going to fool myself trough the help of your economic theories. When reliable technical information is missing, or it is a secret, the information vacuum might be filled with mythology, but this is not what I am asking for. I do know, although it is a kind of secret, that the algorithm of recording, which is implemented in Linux, is that same lossy algorithm. Therefore, you need at least 192 kHz/32-bit to ensure a minimal quality of digital recording. Why 192 kHz/32-bit? This is because of the voodoo of the lossy sampling algorithm and "failure of bandlimitation" which tends to be "referred to as aliasing"http://en.wikipedia.org/wiki/Analog_rec ... g#Aliasinghttp://en.wikipedia.org/wiki/Nyquist%E2 ... iderationshttp://en.wikipedia.org/wiki/Nyquist%E2 ... m#Aliasing
The problem is that the lossy sampling algorithm (which is implemented in Linux for everything, for which it should not be applied) does not satisfy the conditions of the Nyquist theorem
The lossy sampling algorithm of Linux is fundamentally wrong simply because it does not satisfy the conditions of the Nyquist–Shannon sampling theorem.
The [Nyquist] theorem also leads to a formula for reconstruction of the original signal. The constructive proof of the theorem leads to an understanding of the aliasing that can occur when a sampling system does not satisfy the conditions of the [Nyquist] theorem. http://en.wikipedia.org/wiki/Nyquist%E2 ... ng_theorem
That is why I really want to disable any unwanted resampler.
cesium wrote:I had an idea on how to test this but I don't think it would work now (connect output to line-in. Play a low rate output and record it, and than test if the result sounds fine in a low rate. But the recording process's ADC is a sort of SRC too, so I'm not sure how to distinguish...).
It is difficult to believe that it is an unsolvable problem. There is a particular soundcard. There are drivers which work in some way. This means that somebody knows something about the soundcard.