MediaPortal Audio renderer - better video playback quality | Page 96

OK, here it goes. The type of fluctuation that can be seen here comes maybe every 15-20 seconds. One thing though - I tried another ripped HD movie which seemed to play fine. But this one haven't given me any problems before, so...

 

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Have you checked with DPC latency cheker (DPC Latency Checker) if there is some HW / driver issue? Since the graphs have only "one" glitch that appears suddently I would assume it is something external that steals the CPU/GPU time from rendering process.

 

New release available for testing: https://forum.team-mediaportal.com/637138-post3.html

version 18 - based on SVN revision 26672

• Fixed audio delay value - delay was always 10x bigger than it was configured in registry. Remember to update the setting if you have used it (set to non zero value).
• Disabled AC3 encoding when audio stream is either mono or stereo. Improves sound quality when PCM instead of AC3 can be used
• Fixed AC3 drifting calculations - no more jumpy dirft values
• Reduced drift allowed window back to 8 ms with 1:1, 1:2 or 1:3 material

 

Well since it is only certain files that have these problemsI thought it shouldn't be HW related, but I don't know? I let the Latency tester run for a couple of minutes. The maximum latency was 157 (in whatever unit), and the program said that I shouldn't have any problems. I'll run the video while checking CPU load etc as well. I have a high spec PC so it has more than enough grunt.

 

Oh, didn't see it is only few specific files. In that case it shouldn't be EVR renderer nor the audio renderer related issue at all, but instead something before those in the filter chain. Does those "drops" happen in the same place of the file when you play the file again? Or is it just random behavior?

Also try using non-DXVA since ATI has issues with some "scene" encodings for H264. Also ATI has issues with corrupting some specific DVB H264 content - could be related to this as well.

 

Ok, I've been trying this for a while already and tonight took some time to properly set the config.
I have Gigabyte GA-790XTA-UD4 with Realtek ALC889 codec under Windows 7.
With 5.1 analogue sound connected directly.

When I set the audio output settings to output 16 bit integer and enable EnableAC3Encoding using live or recorded TV both stereo and AC3 play perfect.
But when I start a DVD / HD movie the sound doesn't contain the voices of people.

Then if enable pass-trough the sound if perfect but not lip synced, as described.

When I enable 16 bits LPCM the sound is correct and lip synced, but I can't lower the volume with the MP remote.

Is there anything else I can do to get correct sound, synced correctly and using the MP remote?

Thanks in advance.

 

Analog cannot be used for AC3 encoding. AC3 is digital output format. Or does this mean that you have coaxial SPDIF instead of the fiber cable? -> S/PDIF - Wikipedia, the free encyclopedia

If the connection was analog then using AC3 encoding makes MP audio renderer to reject the incoming formats since it cannot output those to WASAPI device (device will reject those). So, what has most likely happened is that directsound default audio renderer was used.

Was this done in ffdshow or other audio codec settings? If yes this will also make MP audio renderer to reject the audio stream. It cannot operate on encoded streams like stated in the 1st post of this thread.

This time most likely the audio renderer was correctly used (logs would help ). Again, the 1st post would have stated that the MP audio renderer is not currently providing volume control support (WASAPI exclusive mode disables all OS level volume control and there is no own implementation yet that would allow the volume level changes).

I would recommend everyone to "find a way" to use the amp's remote to control the volume level since doing it on PC side will reduce the audio stream dynamics and ruins the audio quality (the lower the volume would be set the worser the audio quality would be - this is one of the reasons why I haven't kept any hurry with volume implementation, quality should come first).

 

Thanks for the replies.

The issue with the voices is not related MediaPortal Audio renderer (nor v18)

I use SAF v5 unlocked with the preferred settings. The settings for audio is for Video / TV / DVD all ffshow.

I mean that I connect the analogue output not using SPDIF.
The boxes are directly connected to the PC.

ffdshow indeed.



The DVD and the HD I watch contain AC3 sound only.
If I use 16 integer output, no pass-trough and disable EnableAC3Encoding in register, I get incorrect sound.
I will try with this settings to watch the HD movie and collect the logs.
Tonight I will be home after midnight, so go into my bed directly.
Will collect the logs this weekend.