cesium wrote:hifihere: Does "osstest" work and emit sound? Does 'ossinfo -v3' emit lots of output? If so, oss is installed. Now, I think this wiki page will show you how to configure stuff to use OSS. (KDE, for example uses phonon for sound. Latest gnome IIRC uses libcanberra for notifications). If anything is unclear, please ask here.
cesium wrote:I think this fell by the wayside...
hifihere wrote:So back to my original objective, up to 24/192 support, gapless FLAC playback and bit perfect output.
Maybe I am trying to achieve the immpossible.
cesium wrote:I suggesting leaving Pulse there. It's too difficult to remove given package dependencies, and it's unnecessary really.
type yum erase pulseaudio to remove pulse audio and it's garbage.
igorzwx wrote:Is it possible to get "bit perfect sound" with PulseAudio?
hifihere wrote:Bit perfect is working since I can pass the DTS test. I have to set my sound device to hw or spdif to bypass all the digital manipulation and then it passes bit perfect tests.
If you are running bit-perfect, you will have a more true reproduction from the media that you are playing.
Bit-perfect can only truly be tested using the tos-link out of either your sound card or a compatible audio codec for onboard sound for some motherboards and a DTS-decoder. http://www.mp3car.com/vbulletin/faq-emp ... rfect.html
"TOSLINK cables are widely reported to introduce jitter (essentially timing errors), which results is a less well articulated sound. However, many humans do not notice any resulting distortions. http://en.wikipedia.org/wiki/S/PDIF#Limitations"
S/PDIF lacks flow control and retry facilities, which limits its usefulness in applications outside of synchronous playback.
Because the receiver cannot control the data rate, it instead has to avoid bit slip by synchronising its conversion with the source clock. This means that S/PDIF cannot fully decouple the final signal from influence by the analogue characteristics of the source or the interconnect, even though the digital audio data can normally be transmitted without loss. The source clock may carry inherent jitter or wander, and noise or distortion introduced in the data cable may further influence the process of clock recovery.. If the DAC does not have a stable clock reference then noise will be introduced into the resulting analogue signal. However, receivers can implement various strategies which limit this influence.http://en.wikipedia.org/wiki/S/PDIF#Limitations"
cesium wrote:igorzwx:igorzwx wrote:Is it possible to get "bit perfect sound" with PulseAudio?
I dunno. What I was trying to say/do is this: hifi doesn't want "bit perfect source" for everything, but only for a specific use. I doubt he cares if (for example) the k3b burning success sound is bit perfect or not. This means we could have (assuming OSS install had worked for him and playback was gapless) kept most of the system using Pulse (since reconfiguring everything is work and Pulse can work with OSS if we modify its config file), and migrate only a few important programs (i.e. the FLAC player) to use OSS directly. After that, it's on a base of 'if we care enough, we'll change the output, but we probably don't care'.
Darn. You may wish to follow this, and just build OSS from source**. "make install" should than do.hifihere wrote:I never did get osstest or any OSS commands to work even though it says "already installed" when I try to re-install.
Yeah, I don't favour this.hifihere wrote:I did remove Pulseaudio but my player was broken after that so I reinstalled it.
You mean output to "hw:0" etc.? Can't your player output to ALSA directly? If so, I think setting up the right output address should do..hifihere wrote:Bit perfect is working since I can pass the DTS test. I have to set my sound device to hw or spdif to bypass all the digital manipulation and then it passes bit perfect tests.
I wonder, what makes you think it's not possible?Even if I was able to get OSS4 to work it looks like I have another problem that no one has solved. Gapless playback is not possible and that is a deal breaker for me.
hifihere wrote: The problem that faces all Linux users is chopped up, gapped music playback.
hifihere wrote:I have succeeded in getting Squeezeslave to operate only to find that it can only support 16/44.1 resolution.
So back to my original objective, up to 24/192 support, gapless FLAC playback and bit perfect output.
The Vortexbox/MPD player covers all but gapless FLAC playback. I was hoping that moving to OSS4 would help that problem. I have discovered that others before me have found the MPD player to blame for the rough FLAC playback. It is not an artifact of ALSA/Pulseaudio apparently.
Maybe I am trying to achieve the immpossible.
[slim] Gap of a few seconds in FLAC music playback on my SB1
Mon, 31 Dec 2007 10:30:07 -0800
I posted the following in the Beta forum but have not had any replies
yet. So this afternoon, I decided to uninstall SC7 and downgrade to
SlimServer Version: 6.5.4 - 12568. So having installed version 6.5.4
and let it do a full music scan, I was very disappointed to see that I
am STILL suffering the same problem. Getting a gap of a few seconds
when doing things like searching for music or when the web browser
freshes the music playing screen. The music plays fine with NO gaps if
I am accessing other web sites other than slimserver. I have no problem
searching for music with the remote control, as long as the slimserver
web browser is not active.
I only get this problem with FLAC files. MP3 files play ok all the time
even when the web browser refreshes.
I am currently running SqueezeCenter Version: 7.0 - 15528 on a Windows
Vista Ultimate Dell PC with 3Gb RAM.
My SqueezeBox is an SB1.
This is what I posted to the Beta forum, but is also seems to apply to
6.5.4 too for me!!!!!
I recently upgraded to SC7 and have been suffering a problem when using
the web browser. I control my SB1 with a Dell Axim PDA using Internet
Explorer. When the web browser refreshes the screen each time a new
track starts playing I get a gap of a few seconds in the music
playback. This also happens if I use the web browser on my PC. If I
close down all web browsers and control my SB1 with the remote control,
it works fine.
Is this a problem with my SB1?
Any advice would be appreciated.
Re: [slim] Gap of a few seconds in FLAC music playback on my SB1
Wed, 02 Jan 2008 13:30:07 -0800
I haven't used my SB1 for some time now but from what I can recall the
behaviour you are seeing is quite normal.
The buffer in the SB1 is much smaller than the SB2/3 and is of the
order of a few seconds of music for flac (which will be transcoded to
wav), probably 10 times that for mp3.
This means that if your server is busy for a second or two doing
something else or querying the Slimserver/SqueezeCenter database it's
quite easy to get a dropout. You don't get the same problem with mp3
since the buffer holds 10 to 20 seconds of music.
The remote seemed to use less server power than the web interface and
was less likely to give problems.
I wasn't sure of the numbers above so if you look here:
http://wiki.slimdevices.com/index.cgi?H ... Comparison
you can see the size of the buffers. The numbers there suggest that the
SB1 can only store less than a second of wav data.
Hope this helps.
I have some trouble when playing sound files with Windows Media Player or Media Monkey. When I play music there are very short interruptions. What could be causing this?
My system has 4GB RAM, an Athlon X2 and Soundblaster Audigy soundcard. It runs Windows XP
Прерывается звук в Ubuntu 10.10
« : Сегодня [today] в 16:47:30 »
У меня прерывается звук при воспроизведении любых медиафайлов (аудио, видео, флеш - audio, video, flash) в Ubuntu 10.10. В чём может быть причина?
Intel Corporation 82852/855GM Integrated Graphics Device (rev 02)
Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 03)
cesium wrote:I suggesting leaving Pulse there. It's too difficult to remove given package dependencies, and it's unnecessary really. Also, with some fiddling it can even output to OSS4. You can then migrate programs slowly to OSS if you want to (you may have to edit MPD config files to make sure it chooses OSS output).
cesium wrote:not sure how it's done in Fedora
cesium wrote:Put this file in ~/.pulse directory. You may need to restart Pulse service (not sure how it's done in Fedora) - probably do just "sudo killall -9 pulseaudio" and it (I think) starts again when a program asks for it. Hopefully, this will get Pulse working, and with it most GUI stuff. Afterwards, we'll look into more direct use of OSS. Also, do you want default output to go via SPDIF? If so, does outputting to /dev/dsp emits sound via SPDIF? (You can test with something like "ossplay /usr/share/sounds/*"). If not and you want sound mainly via SPDIF, we may have to do some other changes.
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