i have droputs every second to a few seconds. should i record the audio and send it to you (ogg or mp3)? pd's log window says:
sysRef: 10014 asioversion: 2 driverversion: 11 name: Hercule DJ Console ASIO ASIO initialized successfully ASIOGetChannels inputs: 2, outputs: 6 ASIOGetBufferSize min: 96, max: 1440, preferred: 192, granularity: 96 ASIO: using traditional scheduler ASIOSetSampleRate 48000 ASIO: buffers allocated ASIO: channel 0 type 16 ASIO: channel 1 type 16 ASIO: channel 2 type 16 ASIO: channel 3 type 16 ASIO: input latency: 1440, output latency: 1440 ASIO: ringbuffer size: 2880 ASIO: started
Tim Blechmann wrote:
hi all
to make it easier to debug the native asio implementation, i uploaded precompiled binaries to my website ...
i added a new win32 binary that's using a modified scheduler to compute dsp during the driver callback to reduce latency ...
since i don't have a proper audio device to test it (will only work for hardware buffers of multiples of 64 at the moment), i need any feedback that i can get ...
url: www.geocities.com/themokabar/software.html
thanks a lot ... tim
Hi, do you have some high priority software running apart from PD (like task manager)? Does it become better with higher buffer sizes?
thanks! Thomas
Am 26.11.2004 um 15:40 schrieb smoerk:
i have droputs every second to a few seconds. should i record the audio and send it to you (ogg or mp3)? pd's log window says:
sysRef: 10014 asioversion: 2 driverversion: 11 name: Hercule DJ Console ASIO ASIO initialized successfully ASIOGetChannels inputs: 2, outputs: 6 ASIOGetBufferSize min: 96, max: 1440, preferred: 192, granularity: 96 ASIO: using traditional scheduler ASIOSetSampleRate 48000 ASIO: buffers allocated ASIO: channel 0 type 16 ASIO: channel 1 type 16 ASIO: channel 2 type 16 ASIO: channel 3 type 16 ASIO: input latency: 1440, output latency: 1440 ASIO: ringbuffer size: 2880 ASIO: started
Tim Blechmann wrote:
hi all
to make it easier to debug the native asio implementation, i uploaded precompiled binaries to my website ...
i added a new win32 binary that's using a modified scheduler to compute dsp during the driver callback to reduce latency ... since i don't have a proper audio device to test it (will only work for hardware buffers of multiples of 64 at the moment), i need any feedback that i can get ... url: www.geocities.com/themokabar/software.html thanks a lot ... tim
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nothing running beside Pd, freshly booted windows xp. afaik there is no option in pd to adjust buffer size, but it seems that Pd chose the bigest possible buffer size or what does "ASIO: input latency: 1440, output latency: 1440" mean?
I'm running a Pd patch with an oggcast~ object running that consumes around 20% CPU on a 1.6 GHZ Pentium-M with a USB audio card.
Thomas Grill wrote:
Hi, do you have some high priority software running apart from PD (like task manager)? Does it become better with higher buffer sizes?
thanks! Thomas
Am 26.11.2004 um 15:40 schrieb smoerk:
i have droputs every second to a few seconds. should i record the audio and send it to you (ogg or mp3)? pd's log window says:
sysRef: 10014 asioversion: 2 driverversion: 11 name: Hercule DJ Console ASIO ASIO initialized successfully ASIOGetChannels inputs: 2, outputs: 6 ASIOGetBufferSize min: 96, max: 1440, preferred: 192, granularity: 96 ASIO: using traditional scheduler ASIOSetSampleRate 48000 ASIO: buffers allocated ASIO: channel 0 type 16 ASIO: channel 1 type 16 ASIO: channel 2 type 16 ASIO: channel 3 type 16 ASIO: input latency: 1440, output latency: 1440 ASIO: ringbuffer size: 2880 ASIO: started
i have droputs every second to a few seconds. should i record the audio and send it to you (ogg or mp3)? pd's log window says:
i don't think that's necessary ... i had the same problem earlier, but solved it for my quattro (don't remember how)
ASIOGetBufferSize min: 96, max: 1440, preferred: 192, granularity: 96 ASIO: using traditional scheduler
i changed the asio buffer size to the maximum buffer size ... in your case 1440, that's not a multiple of 64, so you are not using the callback based scheduler ... i'll somehow have to implement a feature to select this, but don't know, where ...
i think, i know, how your clicks sound like, but you can record a short bit and send it to me (privately, and please only a few kb ... slow dialup connection)
i think, there is a bug somewhere in the ringbuffer filling/reading code, but i'm not sure, where ...
i'm playing a concert this evening ... maybe i can look into it tomorrow...
cheers & thanks for testing ... tim
Tim Blechmann wrote:
i have droputs every second to a few seconds. should i record the audio and send it to you (ogg or mp3)? pd's log window says:
i don't think that's necessary ... i had the same problem earlier, but solved it for my quattro (don't remember how)
ASIOGetBufferSize min: 96, max: 1440, preferred: 192, granularity: 96 ASIO: using traditional scheduler
i changed the asio buffer size to the maximum buffer size ... in your case 1440, that's not a multiple of 64, so you are not using the callback based scheduler ... i'll somehow have to implement a feature to select this, but don't know, where ...
and i wondered why it says traditional scheduler :-). maybe you can change the buffer size from maximun to preferred again. is preferred the value you set in the control panel of the audio device or normally a hard coded value in the audio device driver? (i guess this differs from card to card). I think the way to go is to open the ASIO driver's control panel as most audio software does it.
i think, i know, how your clicks sound like, but you can record a short bit and send it to me (privately, and please only a few kb ... slow dialup connection)
just realised that i need am adapter from RCA to mini-jack. but it sounds really like normal dropout, when buffer is to low or computer to slow.
and i wondered why it says traditional scheduler :-). maybe you can change the buffer size from maximun to preferred again. is preferred the value you set in the control panel of the audio device or normally a hard coded value in the audio device driver? (i guess this differs from card to card). I think the way to go is to open the ASIO driver's control panel as most audio software does it.
for most devices i got feedback, the driver control panel sets not only the prefered buffer size, but also minimum and maximum ... your driver is probably different ... i'll use the prefered again ...
cheers ... tim