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Linux AudioEngine testers - Linux only
(2013-01-10, 20:42)uNiversal Wrote: How about you ask there?

Or ask in RPI subforum I doubt many ppl have a RPI to begin with.

uNi

LOL, I haven't tried the RPi subforum here, but I've tried asking at the official rpi forums about stuff and have received no response. But I'll try. Thanks again for listening.
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I compiled XBMC 12.0-RC3 Git:20130110-c79c5d5 yesterday on
Ubuntu Release 13.04 (raring) 64-bit with kernel Linux 3.7.0-7-generic and GNOME 3.6.2.

There was some problem with ./configure. I got this error:
/usr/include/python2.7/Python.h:8:22: fatal error: pyconfig.h: No such file or directory

The file pyconfig.h was missing in the /usr/include/python2.7 folder. However ./configure and make worked when I copied /usr/include/x86_64-linux-gnu/python2.7/pyconfig.h to /usr/include/python2.7/
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hi, I think it's better post here too what I found. my configuration:

- zotac HD-ID11 ION2 > usb > fatman amplifier iTUBE mkII with DAC on board **
- openELEC 3 RC1

the system is also connected via HDMI to Samsung TV

** dac reported with lsusb:
CM102-A+/102S+ Audio Controller
h**p://www.cmedia.com.tw/ProductsDetail.aspx?C1Serno=2&C2Serno=2&C3Serno=5&PSerno=17

first, sound is distorted. volume is very high, as I little turn up on ampli it sounds very loud. music is not clear, fat and compressed, it seems that signal input is kept in clipping.

with the same hardware if I boot in windows 8 instead openeELEC and launch xbmc in wapasi mode > usb, everything is near right. it doesn't reach the excellence of foobar (there's lot of difference, but is OT), but acceptable.

I boot into openeELEC again and sound is horrible, but what is strange is that I lost left channel too!
I thought that was ampli fault but didn't.
I found instead that if I boot openELEC with volume of TV>0 the left channel via usb dac is mute. I tried to reboot with TV volume at 0 and left channel turned on, retried with volume up and it turned off!

this, with the fact that sound is very distorted, make me think that there's some interation across outputs.
but, in fact, I have no idea, I only thought that it could be a kind of thing that you coders would to know; of course I hope that I'll resolve my issues too.
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Have you tried turning down the amplification inside XBMC?

I mean: you presumably have a volume control on your sound system as well as in-xbmc volume.
My system starts clipping if my XBMC is turned up over -25dB. If you set XBMC to 0DB everything clips badly.
System: Kodi on NVidia Shield 2015
Video: Panasonic AE3000 Projector / Samsung 46" LCD
Audio: Quad 2912 on Nord DM500Up with Marantz 7010 receiver.
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I understand, but why is it supposed to be necessary?
and in a bit-perfect system shouldn't be left to 100%?
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Full debug log
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(2013-01-17, 08:18)8rnity Wrote: I understand, but why is it supposed to be necessary?
and in a bit-perfect system shouldn't be left to 100%?

No, because in passthrough mode XBMC does not touch the audio stream at all the receiver gets all of it, so if your outputting distortion you get distortion out no?

However since you have a TV as a audio receiver DTS, AC3 DTS-HD and True-HD and boost volume on downmix should all be disabled. Since TV only handles LPCM at best.

uNi
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(2013-01-17, 09:37)uNiversal Wrote:
(2013-01-17, 08:18)8rnity Wrote: I understand, but why is it supposed to be necessary?
and in a bit-perfect system shouldn't be left to 100%?

No, because in passthrough mode XBMC does not touch the audio stream at all the receiver gets all of it, so if your outputting distortion you get distortion out no?

However since you have a TV as a audio receiver DTS, AC3 DTS-HD and True-HD and boost volume on downmix should all be disabled. Since TV only handles LPCM at best.

uNi
The poster is not using a TV, he is using an external USB DAC.

The DAC is only stereo - so you are right he needs to disable DTS, AC3 passthrough etc. He also needs to give a debug log (wiki). He perhaps also needs to tell us what format he is trying to play (given the audiophile amp, maybe flac? pcm?)

XBMC should be able to pass through stereo digital signals to an external DAC shouldn't it?
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@nickr

Er it may go through the cable, but its not digital when it gets there IIRC the internal DAC Digital to Analog Converter gets to another external DAC as it left the original device.

uNi
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my ampli just has a basic dac internal in same case based on CM102-A+/102S+ Audio Controller that is a single chip dac used most in usb speakers, that has a decent audiophile grade too. it handles 16 bit stereo 44/48K via USB and of course it converts in analogue signal the streaming provided via USB.

dac output is analogic (electrical) and the 2-ch are soldered internally to one of the two line-inputs (the other is RCA and is on the back, there's a switch on front to select wich line-input you want), so I have a pc, an USB cable that connects pc with internal dac of my amplifier, and out of dac it's all done in analogic domain. of course I have a volume knob to regulate the volume of the ampli also (valve pre-amplifier stage).

in xbmc the device is detected as USB speakers and of course I disabled DTS, AC3 passthrough etc.
I feed it with ALAC CDrip files that should be converted in PCM before streaming to external dac via usb.

using XBMCuntu I launched dmix, shouldn't be all at 100% (master volume and board volume) to keep signal untouched?

everything works well with windows wasapi, 100% volume, and foobar as source, even with windows xbmc signal levels are correct and no clipping or distorsion at all (worst than foobar but this is OT).

here some cut of log:

21:31:37 T:139902788302592 DEBUG: NEWADDON PythonCallbackHandler construction with PyThreadState 0x7f3da0000920
21:31:39 T:139905032124224 DEBUG: Previous line repeats 15 times.
21:31:39 T:139905032124224 DEBUG: Joystick 0 button 1 Down
21:31:39 T:139904843585280 DEBUG: CSoftAE::Run - Sink restart flagged
21:31:39 T:139904843585280 INFO: CSoftAE::InternalOpenSink - sink incompatible, re-starting
21:31:39 T:139904843585280 INFO: CAESinkALSA::Initialize - Attempting to open device "@:CARD=Device,DEV=0"
21:31:39 T:139904843585280 INFO: CAESinkALSA::Initialize - Opened device "front:CARD=Device,DEV=0"
21:31:39 T:139904843585280 INFO: CAESinkALSA::InitializeHW - Your hardware does not support AE_FMT_FLOAT, trying other formats
21:31:39 T:139904843585280 INFO: CAESinkALSA::InitializeHW - Using data format AE_FMT_S16NE
21:31:39 T:139904843585280 DEBUG: CAESinkALSA::InitializeHW - Request: periodSize 512, periods 16, bufferSize 8192
21:31:39 T:139904843585280 DEBUG: CAESinkALSA::InitializeHW - Got: periodSize 512, periods 16, bufferSize 8192
21:31:39 T:139904843585280 DEBUG: CAESinkALSA::InitializeHW - Setting timeout to 186 ms
21:31:39 T:139904843585280 DEBUG: CSoftAE::InternalOpenSink - ALSA Initialized:
21:31:39 T:139904843585280 DEBUG: Output Device : USB Sound Device
21:31:39 T:139904843585280 DEBUG: Sample Rate : 44100
21:31:39 T:139904843585280 DEBUG: Sample Format : AE_FMT_S16NE
21:31:39 T:139904843585280 DEBUG: Channel Count : 2
21:31:39 T:139904843585280 DEBUG: Channel Layout: FL,FR
21:31:39 T:139904843585280 DEBUG: Frames : 512
21:31:39 T:139904843585280 DEBUG: Frame Samples : 1024
21:31:39 T:139904843585280 DEBUG: Frame Size : 4
21:31:39 T:139904843585280 DEBUG: CSoftAE::InternalOpenSink - Using speaker layout: 2.0
21:31:39 T:139904843585280 DEBUG: CSoftAE::InternalOpenSink - Internal Buffer Size: 4096
21:31:39 T:139904843585280 DEBUG: AERemap: Downmix normalization is enabled
21:31:39 T:139904843585280 DEBUG: CSoftAEStream::CSoftAEStream - Converting from AE_FMT_S16NE to AE_FMT_FLOAT
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I have a funny problem with the new AE in frodo RC3.
I use optical spdif of my Club3D Theatron DD for both passthrough and audio. Passthrough works fine (movies), but when playing music (mp3, flac) I don't have any sound.
All other audio devices work just fine (onboard optical, hdmi) only my theatron doesn't...
Any suggestions?

I use Ubuntu 12.04 64bit.
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(2013-01-17, 10:40)uNiversal Wrote: @nickr

Er it may go through the cable, but its not digital when it gets there IIRC the internal DAC Digital to Analog Converter gets to another external DAC as it left the original device.

uNi

What internal dac?
(2013-01-17, 14:36)8rnity Wrote: my ampli just has a basic dac internal in same case based on CM102-A+/102S+ Audio Controller that is a single chip dac used most in usb speakers, that has a decent audiophile grade too. it handles 16 bit stereo 44/48K via USB and of course it converts in analogue signal the streaming provided via USB.

dac output is analogic (electrical) and the 2-ch are soldered internally to one of the two line-inputs (the other is RCA and is on the back, there's a switch on front to select wich line-input you want), so I have a pc, an USB cable that connects pc with internal dac of my amplifier, and out of dac it's all done in analogic domain. of course I have a volume knob to regulate the volume of the ampli also (valve pre-amplifier stage).

in xbmc the device is detected as USB speakers and of course I disabled DTS, AC3 passthrough etc.
I feed it with ALAC CDrip files that should be converted in PCM before streaming to external dac via usb.

using XBMCuntu I launched dmix, shouldn't be all at 100% (master volume and board volume) to keep signal untouched?

everything works well with windows wasapi, 100% volume, and foobar as source, even with windows xbmc signal levels are correct and no clipping or distorsion at all (worst than foobar but this is OT).

here some cut of log:

21:31:37 T:139902788302592 DEBUG: NEWADDON PythonCallbackHandler construction with PyThreadState 0x7f3da0000920
21:31:39 T:139905032124224 DEBUG: Previous line repeats 15 times.
21:31:39 T:139905032124224 DEBUG: Joystick 0 button 1 Down
21:31:39 T:139904843585280 DEBUG: CSoftAE::Run - Sink restart flagged
21:31:39 T:139904843585280 INFO: CSoftAE::InternalOpenSink - sink incompatible, re-starting
21:31:39 T:139904843585280 INFO: CAESinkALSA::Initialize - Attempting to open device "@:CARD=Device,DEV=0"
21:31:39 T:139904843585280 INFO: CAESinkALSA::Initialize - Opened device "front:CARD=Device,DEV=0"
21:31:39 T:139904843585280 INFO: CAESinkALSA::InitializeHW - Your hardware does not support AE_FMT_FLOAT, trying other formats
21:31:39 T:139904843585280 INFO: CAESinkALSA::InitializeHW - Using data format AE_FMT_S16NE
21:31:39 T:139904843585280 DEBUG: CAESinkALSA::InitializeHW - Request: periodSize 512, periods 16, bufferSize 8192
21:31:39 T:139904843585280 DEBUG: CAESinkALSA::InitializeHW - Got: periodSize 512, periods 16, bufferSize 8192
21:31:39 T:139904843585280 DEBUG: CAESinkALSA::InitializeHW - Setting timeout to 186 ms
21:31:39 T:139904843585280 DEBUG: CSoftAE::InternalOpenSink - ALSA Initialized:
21:31:39 T:139904843585280 DEBUG: Output Device : USB Sound Device
21:31:39 T:139904843585280 DEBUG: Sample Rate : 44100
21:31:39 T:139904843585280 DEBUG: Sample Format : AE_FMT_S16NE
21:31:39 T:139904843585280 DEBUG: Channel Count : 2
21:31:39 T:139904843585280 DEBUG: Channel Layout: FL,FR
21:31:39 T:139904843585280 DEBUG: Frames : 512
21:31:39 T:139904843585280 DEBUG: Frame Samples : 1024
21:31:39 T:139904843585280 DEBUG: Frame Size : 4
21:31:39 T:139904843585280 DEBUG: CSoftAE::InternalOpenSink - Using speaker layout: 2.0
21:31:39 T:139904843585280 DEBUG: CSoftAE::InternalOpenSink - Internal Buffer Size: 4096
21:31:39 T:139904843585280 DEBUG: AERemap: Downmix normalization is enabled
21:31:39 T:139904843585280 DEBUG: CSoftAEStream::CSoftAEStream - Converting from AE_FMT_S16NE to AE_FMT_FLOAT

Please post the entire log to pastebin
If I have helped you or increased your knowledge, click the 'thumbs up' button to give thanks :) (People with less than 20 posts won't see the "thumbs up" button.)
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@nickr

Every Audio card system has a dac onchip IIRC its necessary!

How else you think sound is processed internally? magic?

uNi
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With respect to using an external USB DAC it can be something like this: http://www.cambridgeaudio.com/products/d...-converter

USB input from PC (likely bitstreamed LPCM from CD / FLAC source in case of music; will be massaged by XBMC if other source) --> DAC --> Analog to stereo.

In this case there is no Digital-Analog conversion done in the PC...point is to bypass the cheapo DAC & noisy environment in the PC and have a higher quality DAC do it.

Most of these units can also optionally 'upconvert' the input signal too, but main point is to not use the $2 DAC in the PC. I plan to do this on my Rotel + B&W setup one day...those kids cartoons will never sound the same!

If I helped out pls give me a +

A bunch of XBMC instances, big-ass screen in the basement + a 20TB FreeBSD, ZFS server.
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(2013-01-17, 21:02)uNiversal Wrote: @nickr

Every Audio card system has a dac onchip IIRC its necessary!

How else you think sound is processed internally? magic?

uNi

Uni there is only one DAC here, it is in the amp. So why your post about 2 DACs is meaningless.
If I have helped you or increased your knowledge, click the 'thumbs up' button to give thanks :) (People with less than 20 posts won't see the "thumbs up" button.)
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