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Old 15th April 2007, 05:32 PM   #121
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Quote:
Originally posted by peufeu
Anyway I still believe you'd be better off with slaving the PC to the DAC and remove the problem
This is of course the major problem with the way they fudged USB audio to work.

Somehow I doubt you actually have a 1 sample buffer, with an scheduler period of 0.5ms you've got 24 stereo samples being played out between each scheduler interrupt.
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Old 15th April 2007, 07:15 PM   #122
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Quote:
Originally posted by BlackCatSound


This is of course the major problem with the way they fudged USB audio to work.

Somehow I doubt you actually have a 1 sample buffer, with an scheduler period of 0.5ms you've got 24 stereo samples being played out between each scheduler interrupt.
One sample is that what I configured with ALSA. (Though I didn't measure it. )
I am not sure if e.g. the USB-controller or the USB-snd-driver buffers extra samples.

The PCM2707 buffers 3 stereo samples on its own.
In case another IRQ pops in you'd be right. I'd run into an underrun. But that's
not happening. There is a slight chance that there is another big buffer sitting somewhere. (Or no other IRQs are bothering me since mainly everthing is switched
off in the system and the system remains untouched while playing back)

People without rt-scheduling won't make it below 32samples ALSA buffer.

One thing for sure. There are still plenty of dark holes in a PC envrironment messing around with the stream. My intention is still to get these under control as much as possible. But it takes hell lot of work and knowledge.
The next action would be then to analyse the usb-snd-driver to make sure nothing goes wrong there. First you need to understand the driver itself than you need in-depth knowledge of the USB controler and management asf. All these things are
little projects on its own. peufeu knows what I am talking about.


Cheers
\Klaus



Cheers
\Klaus
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Old 15th April 2007, 08:05 PM   #123
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According to the datasheet the PCM2707 stores 1ms of audio.

Are you still running a dddac?
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Old 15th April 2007, 08:42 PM   #124
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Quote:
Originally posted by BlackCatSound
According to the datasheet the PCM2707 stores 1ms of audio.

Are you still running a dddac?
Yes I do. Somehow I must have mixed that up. But than it's all clear, isn't it! THX
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Old 17th April 2007, 12:18 PM   #125
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The timer at 5000Hz rules! Cheers \Klaus
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Old 17th April 2007, 12:27 PM   #126
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Are you running the dddac in reclock mode?
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Old 17th April 2007, 01:27 PM   #127
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Quote:
Originally posted by BlackCatSound
Are you running the dddac in reclock mode?
I am running an external 12Mhz Tent clock. If that's what you mean.
There is no I2S reclocking.

Perhaps I should mention that my "DDDAC" is heavily tweaked (parts, construction, cabling and battery!)


BTW: I was in contact with Con Kolivas. He pretty much supported the approach to go towards 5000Hz.
He mentioned , that above 5000Hz, you might encounter problems that processes just die. (I'll try it anyhow!)
Beside that he mentioned that the 1ms slices at 1000Hz are not
really 1ms all the time. They can be up to 2-3ms long.
By running higher freqencies I am lowering this non-linearity.

Cheers
\Klaus
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Old 17th April 2007, 07:36 PM   #128
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I am searching for a supplier for a PCM2706 or 2707. My regular supplier asks $20,- for one chip, and it costs only 7,-
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Old 17th April 2007, 08:05 PM   #129
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Tried getting samples from TI?
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Old 18th April 2007, 11:50 AM   #130
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hi Klaus,

Had a quick look at the Kovalis mailing list to follow the audio discussion you instigated there. Interesting that he revealed himself as an audiophile and sounds like the timer could be the key element to the ideal audio setup. Glad he said that sheduler choice may not be that important as some of the discussion over there between the developers was pretty fierce - the level of competiton between them will ensure continued true open source development, but looks like a lot of releases and subsequent recompiling to keep up.

seems like you've had a major breakthrough here.

Anyway I had a google for os timer and audio and one of the first hits was an interesting microsoft article http://www.microsoft.com/whdc/system/CEC/mm-timer.mspx

some useful comment on how developers use buffers to get round the usually insufficiently low frequency timer, and that these buffers are not an optimum solution. Again discussion is based around video and audio synchronisation, but the kovalis comments may suggest audio quality itself is also a parameter.
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