AudioEngine branch - DO NOT REQUEST BINARY BUILDS - Printable Version
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- meridius - 2012-01-17 14:50
hi all, about the audio sync and 24hz frames per sec.
here is some test files which run at 23.976 which has both AC3 and DTS tracks.
I have to delay the audio in DTS tracks to -200ms but i am sure it could be -210ms but XBMC does not let you pick that option.
What i did was run the test videos above with high volume from my amp and then recorded the picture and audio at the same time using my ipad camera app.
I loaded it into imovie (which you can see picture and audio track at the same time in the same time line) on the ipad and i moved the line so it was bang on the highest peek of the audio track and then looked at the video to see if it hit centre and i did this a few times at different delays in the audio.
the closest i could get to the centre was -200m but i am sure it could be -210ms.
I hope this helps people and its a good test to see if your audio is in sync i have not tried the AC3 track yet which would be stereo or dolby digital.
I get no sync problems when i run everything in 60hz but when i use the option that syncs the tv to the frame rate like 24hz i get audio sync problems.
sorry for the quick responce but at work on my iphone and its a pain to type on , lol
hope this helps
- liquidskin76 - 2012-01-17 15:14
Cheers meridius... was about to upload!
- rodercot - 2012-01-17 17:48
gnif Wrote:@rodercot - do NOT tell people to use sudo to compile, your running the compile as root, very very bad juju, a mistake in the make file could execute an "rm / -Rf" and there is nothing to stop it from succeeding.
AGREED and that is not what it was meant as, I was asking why it was needed with this build. I have never used it before and have only had to use when compiling this branch.
I built davide's branch without sudo FYI.
- DDDamian - 2012-01-17 17:50
Hack_kid Wrote:my Onkyo TX-NR609 supports outputing 1 source to botuh zones at the same time but only analog connections it wont do it with the HDMI audio (most AVRs are this way now a days) so to get it to work u simply connect audio to the analog autio in of whatever input your on@Hack_kid I might have mentioned this in another thread, but most AVRs having that feature will allow you to connect via analog patch cables from PRE-OUT -> ZONE 2 INPUT. This will copy your HDMI input from the HTPC to your Zone 2 and do exactly what you want. You may need to enable Zone 2 in the setup of your amp.
Sorry to go off-topic, I know this thread needs to be brought back to the original intent, just know that this is gonna come up again and again.
- Drae - 2012-01-17 18:25
OT still - sorry Gnif.
rodercot Wrote:AGREED and that is not what it was meant as, I was asking why it was needed with this build. I have never used it before and have only had to use when compiling this branch.It's not needed - mine works fine. Most likely you built it in the past with sudo or cloned/pulled with sudo and thus some files are now owned by root. Do a chown -R xbmc.xbmc on the folder containing the source (not in the folder itself - on the folder itself) - obviously replace xbmc.xbmc with whoever should own those files. If it still fails - do a fresh clone.
- joethefox - 2012-01-17 21:03
gnif Wrote:I would also like to know if people are noticing an audible difference with running AE... I am by no means an audiophile, I am only doing this for fun and to fully take advantage of my HiFi equipment.
- DDDamian - 2012-01-17 21:19
joethefox Wrote:Absolutely yes.
I'll get lacerated here for not performing a double-blind study after carefully prepping the files and painting my sound room with rubber, but absolutely yes. I'm using Foobar 2K with WASAPI exclusive as a reference and 24-bit DVD-A rips to FLAC as a source, and the difference between them is now negligible.
If I compare 16-bit XBMC with XBMC-AE it's night-and-day: the clarity, precision and "air" of high-def comes out as it should, just as with Foobar and getting comparable to my standalone DVDA/SACD player over Denon's Link III, which elimates clock jitter unlike HDMI.
I took it a bit further though and ran the audio through as RAW to avoid the integer->float->integer conversion. It's *extremely* unlikely I would have heard ANY difference as a result of that, but it does keep the stream bit-for-bit. Relevent or not it's something someone's gonna ask for
Built your work from last night - gonna test!
- dado483 - 2012-01-17 21:53
mylle Wrote:Hi Davide,
It's very strange......under Linux i don't have any problems.
Is not possible to know the line in the code where there error is?
- gnif - 2012-01-18 01:23
DDDamian Wrote:I took it a bit further though and ran the audio through as RAW to avoid the integer->float->integer conversion.
The precision of float is greater then the precision of 24bit integer, there is NO loss at all when converting between these formats, even bit-streams in pass-through would work via this conversion.
Thanks for testing though, good to know it is making a difference.
- DDDamian - 2012-01-18 01:43
gnif Wrote:The precision of float is greater then the precision of 24bit integer, there is NO loss at all when converting between these formats, even bit-streams in pass-through would work via this conversion.
Agreed, just an absolute purist would complain. FYI you are adding a signal-to-noise ratio of approximately 30,000,000:1 with the conversion lol.
Couple of things:
1) Testing with an ATI HD6470M - supports 16, 24bit from 8khz to 192khz, 8 channels plus all HD formats. I noted that with the code as-is the ATI drivers would not allow exclusive. Made numerous changes to the wfxex initial data and changed the data format to WAVE_FMT_EXTENSIBLE, that got it to work. Every field in the structure needed a valid argument. With your code as-is RealTek drivers for the same card worked fine in Exclusive.
2) I note that once you have a functioning channel layout it does not get changed if the stream format changes. In the mods I had made I had it so that, in Exclusive mode, if the incoming channel layout didn't match then I would re-open the sink with the new layout. E.G. currently going from a six-channel PCM stream to a two-channel stream the sink stays at six-channel, prohibiting AVR-based processing on the two-channel stream.
Unfortunately I had to change CSoftAE::OpenSink() to CSoftAE::OpenSink(AEAudioFormat dataFormat) and pass it the new format data to test for a change in channel count, bitdepth, samplerate etc to avoid the masterStream override you're using. Still, an improvement.
And BTW, welcome the f*&k back!!!!!