Netherlands
- Thread starter
- #41
Odd... You can try to increase the "Playback buffer size" setting to its max of 3 secs and try again. If it still doesn't work, pls post the Mediaportal.log file of that session (loglevel debug).
Sieds
ASIO Music Player Plugin (1 Viewer)
- Thread starter Symphy
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Odd... You can try to increase the "Playback buffer size" setting to its max of 3 secs and try again. If it still doesn't work, pls post the Mediaportal.log file of that session (loglevel debug).
Sieds
Thx Symphy, setting the "Playback buffer size" to 3 secs solved the problem. But if I change any of the settings in the General tab, I become the same problem again. That's no big problem for me. I will now enjoy the sweet tunes created by your ASIO plugin and leave all settings to default. If you are interested investigating this problem, I can create the log for you.
Greetings from a happy guy
Torpedo
Greetings from a happy guy
Torpedo
- Thread starter
- #43
Could it be you have a slow machine or one that's under very heavy load, or play from a slow network? It looks like it takes a very long time to open the next file to be played. The settings on the general tab should not have any effect on this though...
The log, with the failures inthere, might tell us more, so pls create it anyway.
Regards,
Sieds
- December 25, 2004
- 31
- 0
pureaudio do not automatically skip to next song!
yes the tittle says it all! the plugin works well compared to the build in - but I have now once again installed the winamp player and using that WITH an ASIO plugin to get "god sound quality" or also called Bitcorrect audio!
BUT I would prefer to use the more "build in" PureAudio plugin! But it doesnt seem to skip to the next number after every end song... and it doesnt matter if I add songs to the playlist and play from there... the same s... where it simply stops after the last second off music...
Can the programmer off this great PureAudio Plugin look into this issue! I have now installed the MP on 3 machines and the same everytime...
yes the tittle says it all! the plugin works well compared to the build in - but I have now once again installed the winamp player and using that WITH an ASIO plugin to get "god sound quality" or also called Bitcorrect audio!
BUT I would prefer to use the more "build in" PureAudio plugin! But it doesnt seem to skip to the next number after every end song... and it doesnt matter if I add songs to the playlist and play from there... the same s... where it simply stops after the last second off music...
Can the programmer off this great PureAudio Plugin look into this issue! I have now installed the MP on 3 machines and the same everytime...
I stumbled across this plugin a few days ago and installed it immediately. It works! Finally I can now play my DTS files. Thank you, thank you, thank you!
I do however also have the problem of a hickup shortly before the end of a track and I am sure that it has nothing to do with lack of performance. It is a brand new dedicated HTPC machine (AMD 4200+) that only runs MP, so no other processes that are doing anything. CPU usage is very low and there is very little disc activity. My music files are on a local disk so I doubt it is a network issue. I'll try the 3 second buffer setting when I get home tonight.
Symph, is it possible for to you detect if the audiostream is a dts stream? Maybe it would be an idea to automatically switch off the fade in/fade out option when playing a dts file. Because of the fade-in the first 500 msec of a dts file is still noise. Only once the fading has reached 100% it is a bit perfect stream again and the receiver will recognise the stream as being DTS. For now I have just turned off fade-in/fade-out all together.
I do however also have the problem of a hickup shortly before the end of a track and I am sure that it has nothing to do with lack of performance. It is a brand new dedicated HTPC machine (AMD 4200+) that only runs MP, so no other processes that are doing anything. CPU usage is very low and there is very little disc activity. My music files are on a local disk so I doubt it is a network issue. I'll try the 3 second buffer setting when I get home tonight.
Symph, is it possible for to you detect if the audiostream is a dts stream? Maybe it would be an idea to automatically switch off the fade in/fade out option when playing a dts file. Because of the fade-in the first 500 msec of a dts file is still noise. Only once the fading has reached 100% it is a bit perfect stream again and the receiver will recognise the stream as being DTS. For now I have just turned off fade-in/fade-out all together.
- Thread starter
- #46
spraycleaner,
pls try to set the "Playback buffer size" setting to its max of 3 secs. If you still have problems, i really need logfiles (debuglevel) before i can say anything usefull about it.
Regards,
Sieds
synthdood, could you also privide me with logfiles? I might be able to draw conclusions from it. I agree with you that machines like that cannot be the problem. I personally use a Celeron 2.4 (launching MP takes forever on it) and it plays rocksolid over asio.
I will give it some more thoughts but right now i don't have a clue what this problem might be...
Actually i'm currently working on that. The idea is to switch to a "passthrough" mode that disables everything that's in the way, including fading and DSP plugings that are going to be supported in the near future.
The next version will have a feature like that.
I can also announce that the next version will have visualizations!
regards,
Sieds
I was going to produce some logfiles for you, but I had configured the buffer size to 3 sec and the hickup problem has gone away. I tried setting it back to 2 seconds, but the problem did not re-appear.
While I am writing this I just thought of something: I experienced the problem when using the analog output but when I moved the HTPC to my home office to do some tests and extra configuring I connected it through spdif. Could it be that the hickup only happens on the analog output?
While I am writing this I just thought of something: I experienced the problem when using the analog output but when I moved the HTPC to my home office to do some tests and extra configuring I connected it through spdif. Could it be that the hickup only happens on the analog output?
- Thread starter
- #48
I can't think of any reason why it should...
The only thing we know now is that performance is not the issue. Currently i'm testing the upcoming version with visualizations added. Even when i enable the status-overlay which pushes my slow celeron to its limits the ASIO playback doens't get affected in any way...
@synthdood, What kind of device do you use? ASIO4ALL? maybe it's related to particular devices.
Symph
Hello Grubi (if you are still monitoring)
I did as you suggested and picked up an audio card with a C-MEDIA 8738/8768 chip. A TrerraTek with an 8768 chip. Installed the WinRT variant of the driver at http://cmediadrivers.googlepages.com. Uninstalled ASIO4All. Configured PureAudio to output to the SPDIF directly but still nothing but static, clicks and pops. It does fade in and out nicely but no DTS.
The driver you suggested DOES work. I get perfect playback of MP3s but I think that Windows is still getting in the way. If I set Vista to output at 96.0 kHz the the MP3s are up-scaled to this bitrate just not bit perfect.
Any suggestions?
Desperately seeking bit-perfect!
Chris
I did as you suggested and picked up an audio card with a C-MEDIA 8738/8768 chip. A TrerraTek with an 8768 chip. Installed the WinRT variant of the driver at http://cmediadrivers.googlepages.com. Uninstalled ASIO4All. Configured PureAudio to output to the SPDIF directly but still nothing but static, clicks and pops. It does fade in and out nicely but no DTS.
The driver you suggested DOES work. I get perfect playback of MP3s but I think that Windows is still getting in the way. If I set Vista to output at 96.0 kHz the the MP3s are up-scaled to this bitrate just not bit perfect.
Any suggestions?
Desperately seeking bit-perfect!
Chris
I have no experience with Vista at all. On XP I simply installed the normal (non WaveRT driver) selected the output device in PureAuido and all is perfect. Did not have a single issue with it for four month now. Bitperfect 44.1 Khz PCM and DTS output and also paththrough of 48 khz DTS and DD.
grubi.
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