MediaPortal Audio renderer - better video playback quality (2 Viewers)

tourettes

Retired Team Member
  • Premium Supporter
  • January 7, 2005
    17,301
    4,800
    Re: AW: MediaPortal Audio renderer - better video playback quality

    @tourettes

    You have mail.

    Yep, the issue seems to be reproduceble. I'll try to have some time this evening to look at it (my bet is on that the muxing is done really badly and that causes us to drop data. Alter all the samples are sent up to 19 seconds ahead!

    25-07-2010 10:04:45.259 [ 3f8] rtTime: 70258.257 ms rtSampleTime: 89343.965 ms diff -19085.708 ms
    25-07-2010 10:04:45.299 [ 3f8] rtTime: 70299.895 ms rtSampleTime: 89429.298 ms diff -19129.404 ms
    25-07-2010 10:04:45.380 [ 3f8] rtTime: 70384.223 ms rtSampleTime: 89514.632 ms diff -19130.408 ms
    25-07-2010 10:04:45.381 [ 3f8] rtTime: 70385.267 ms rtSampleTime: 89599.965 ms diff -19214.698 ms

    I bet our buffering is not big enough tp handle such. Remuxing the file properly would fix the issue most likely as well :)
     

    te3hpurp

    Retired Team Member
  • Premium Supporter
  • September 23, 2008
    910
    231
    Rovaniemi
    Home Country
    Finland Finland
    Hi. I tried to test, but i'm unable to get to work. I checked it with graphstudio and audiorenderer is not in use.

    Log:
    25-07-2010 07:15:08.060 [ 174c] SetBias: 1.0000000000
    25-07-2010 07:15:08.060 [ 174c] SetBias - updated SoundTouch tempo
    25-07-2010 07:15:08.060 [ 174c] Run
    25-07-2010 07:15:08.060 [ 174c] CheckAudioClient
    25-07-2010 07:15:08.099 [ 174c] Pause
    25-07-2010 07:15:08.127 [ 174c] Stop
    25-07-2010 07:15:08.127 [ 174c] Stop - releasing WASAPI resources
    25-07-2010 07:15:08.187 [ 174c] MP Audio Renderer - v0.61 - instance 0x7c6cf10
    25-07-2010 07:15:08.187 [ 174c] Loading settings from registry
    25-07-2010 07:15:08.187 [ 174c] ForceDirectSound: 0
    25-07-2010 07:15:08.187 [ 174c] EnableTimestrecthing: 1
    25-07-2010 07:15:08.187 [ 174c] WASAPIExclusive: 1
    25-07-2010 07:15:08.187 [ 174c] LogSampleTimes: 0
    25-07-2010 07:15:08.187 [ 174c] DevicePeriod: 500000 (1 == minimal, 0 == default, other user defined)
    25-07-2010 07:15:08.187 [ 174c] WASAPIPreferredDevice:
    25-07-2010 07:15:08.187 [ 174c] GetAvailableAudioDevices
    25-07-2010 07:15:08.194 [ 174c] Audio endpoint 0: "Digital Audio (S/PDIF) (High Definition Audio Device)" ({0.0.0.00000000}.{1cd8e6b9-145a-4d2b-a791-2254ae9db5f2})
    25-07-2010 07:15:08.195 [ 5fc] Render thread - starting up - thread ID: 1532
    25-07-2010 07:15:08.198 [ 70c] Resampler thread - starting up - thread ID: 1804
    25-07-2010 07:15:09.034 [ 174c] Pause
    25-07-2010 07:15:09.741 [ 17a0] SetBias: 1.0000000000
    25-07-2010 07:15:09.741 [ 17a0] SetBias - updated SoundTouch tempo
    25-07-2010 07:15:09.741 [ 17a0] Run
    25-07-2010 07:15:09.741 [ 17a0] CheckAudioClient
    25-07-2010 07:15:09.741 [ 17a0] GetAudioDevice
    25-07-2010 07:15:09.741 [ 17a0] Target end point:
    25-07-2010 07:15:09.741 [ 17a0] GetAvailableAudioDevices
    25-07-2010 07:15:09.747 [ 17a0] Unable to find selected audio device, using the default end point!
    25-07-2010 07:15:09.750 [ 13cc] SetBias: 1.0409559751
    25-07-2010 07:15:09.750 [ 13cc] SetBias - updated SoundTouch tempo
    25-07-2010 07:15:09.750 [ 17a0] CreateAudioClient
    25-07-2010 07:15:09.750 [ 17a0] CreateAudioClient success
    25-07-2010 07:15:47.270 [ 174c] Pause
    25-07-2010 07:15:47.495 [ 174c] SetBias: 1.0000000000
    25-07-2010 07:15:47.495 [ 174c] SetBias - updated SoundTouch tempo
    25-07-2010 07:15:47.495 [ 174c] Run
    25-07-2010 07:15:47.495 [ 174c] CheckAudioClient
    25-07-2010 07:15:59.261 [ 13a8] Pause
    25-07-2010 07:16:12.137 [ 174c] MP Audio Renderer - destructor - instance 0x7c6a4c0
    25-07-2010 07:16:12.137 [ 174c] Stop
    25-07-2010 07:16:12.137 [ 16e8] Resampler thread - closing down - thread ID: 5864
    25-07-2010 07:16:12.137 [ 5d8] Render thread - closing down - thread ID: 1496
    25-07-2010 07:16:12.137 [ 174c] MP Audio Renderer - destructor - instance 0x7c6a4c0 - end
    25-07-2010 07:16:21.722 [ 13a8] SetBias: 1.0000000000
    25-07-2010 07:16:21.722 [ 13a8] SetBias - updated SoundTouch tempo
    25-07-2010 07:16:21.722 [ 13a8] Run
    25-07-2010 07:16:21.722 [ 13a8] CheckAudioClient
    25-07-2010 07:16:36.976 [ 174c] Pause
    25-07-2010 07:16:36.990 [ 174c] Stop
    25-07-2010 07:16:36.990 [ 174c] Stop - releasing WASAPI resources

    So endpoint cannot be found.

    I have Realtek mbo built-in soundchip. I also tested with mpa codec wirh spdif disabled and 16 bit pcm. No-luck.
    Im suspecting Realtek driver.

    Br,
     

    tourettes

    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    17,301
    4,800
    Unable to find selected audio device, using the default end point!

    That is ok. It is just badly worded log text. It just tells that no audio end point was selected
    and a default one will be used.

    I have Realtek mbo built-in soundchip. I also tested with mpa codec wirh spdif disabled and 16 bit pcm. No-luck.
    Im suspecting Realtek driver.

    I would suspect something else, it was not even tried to negotiate the connect between audio decoder and audio renderer. Could be something with the MPA and its output pin's properties that causes the connection to be rejected quite early stage so that there won't be even logging (normaly it would spam alot if all connection tries are logged... even non-audio stuff is tried to connect to the audio renderer :)).
     

    tourettes

    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    17,301
    4,800
    Re: AW: MediaPortal Audio renderer - better video playback quality

    @tourettes

    You have mail.

    Yep, the issue seems to be reproduceble. I'll try to have some time this evening to look at it (my bet is on that the muxing is done really badly and that causes us to drop data. Alter all the samples are sent up to 19 seconds ahead!

    25-07-2010 10:04:45.259 [ 3f8] rtTime: 70258.257 ms rtSampleTime: 89343.965 ms diff -19085.708 ms
    25-07-2010 10:04:45.299 [ 3f8] rtTime: 70299.895 ms rtSampleTime: 89429.298 ms diff -19129.404 ms
    25-07-2010 10:04:45.380 [ 3f8] rtTime: 70384.223 ms rtSampleTime: 89514.632 ms diff -19130.408 ms
    25-07-2010 10:04:45.381 [ 3f8] rtTime: 70385.267 ms rtSampleTime: 89599.965 ms diff -19214.698 ms

    I bet our buffering is not big enough tp handle such. Remuxing the file properly would fix the issue most likely as well :)

    Here we seem to run out of the input data from Haali splitter. It wont happen if resampling is disabled. Also it doesn't seem to be the issue with Reclock. But I'm quite clueless what we could do since we just aren't receiving samples...
     

    Owlsroost

    Retired Team Member
  • Premium Supporter
  • October 28, 2008
    5,540
    5,038
    Cambridge
    Home Country
    United Kingdom United Kingdom
    Re: AW: MediaPortal Audio renderer - better video playback quality

    One thing that can be done is to reduce the drifting (hopefully removing it completely) is to disable the sync correction when bias is 1.0 and refresh rate and source materials fps dont match, since now the EVR presenter is telling to audio renderet that it should adjust the audio resampling a bit, but those are just not working since the major (bias) is not even near the correct speed.

    I'll incorporate a mod for this next time the dshowhelper SVN is updated - just need to test it (I have a sample of a 24Hz-video-in-a-60Hz-wrapper video file from Mironicus already).

    Tony
     

    tourettes

    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    17,301
    4,800
    Re: AW: MediaPortal Audio renderer - better video playback quality

    One thing that can be done is to reduce the drifting (hopefully removing it completely) is to disable the sync correction when bias is 1.0 and refresh rate and source materials fps dont match, since now the EVR presenter is telling to audio renderet that it should adjust the audio resampling a bit, but those are just not working since the major (bias) is not even near the correct speed.

    I'll incorporate a mod for this next time the dshowhelper SVN is updated - just need to test it (I have a sample of a 24Hz-video-in-a-60Hz-wrapper video file from Mironicus already).

    Change it self should be quite minimal.
     

    tourettes

    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    17,301
    4,800
    Re: AW: MediaPortal Audio renderer - better video playback quality

    @tourettes

    You have mail.

    Yep, the issue seems to be reproduceble. I'll try to have some time this evening to look at it (my bet is on that the muxing is done really badly and that causes us to drop data. Alter all the samples are sent up to 19 seconds ahead!

    25-07-2010 10:04:45.259 [ 3f8] rtTime: 70258.257 ms rtSampleTime: 89343.965 ms diff -19085.708 ms
    25-07-2010 10:04:45.299 [ 3f8] rtTime: 70299.895 ms rtSampleTime: 89429.298 ms diff -19129.404 ms
    25-07-2010 10:04:45.380 [ 3f8] rtTime: 70384.223 ms rtSampleTime: 89514.632 ms diff -19130.408 ms
    25-07-2010 10:04:45.381 [ 3f8] rtTime: 70385.267 ms rtSampleTime: 89599.965 ms diff -19214.698 ms

    I bet our buffering is not big enough tp handle such. Remuxing the file properly would fix the issue most likely as well :)

    Here we seem to run out of the input data from Haali splitter. It wont happen if resampling is disabled. Also it doesn't seem to be the issue with Reclock. But I'm quite clueless what we could do since we just aren't receiving samples...

    Ok, found the issue. Next version of the renderer will work those specific files. In the future we might need tweak some of the internal buffer sizes a bit more since it is quite hard to implement those as dynamic ones (it is almost too late when we know the sample sizes... ).
     

    tourettes

    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    17,301
    4,800
    Hi. I tried to test, but i'm unable to get to work. I checked it with graphstudio and audiorenderer is not in use.

    I tested a TS file with MPA audio decoder. It works nicely with the audio renderer. Could you please upload a sample file.
     

    rene.z

    Portal Member
    January 2, 2007
    18
    0
    Home Country
    Austria Austria
    Hello all,

    normally I don't do what I need to do now, however I am quite lost... I have tried to follow (and understand) what you guys are discussing but there is apparently too much background knowledge that I am missing. Here is my problem:

    I am on a Client/Server setting with an Eeebox as client machine with Windows 7, MP 1.1 and SAF 5.0.
    Although liveTV looks nice I do get short "pauses" every couple of seconds (say 20 seconds or so) then MP seems to re-sync and sound and video is nice for another 20 seconds.

    My TsReader log shows alternating "Demux : Audio to render 0.xxx sec" and "Vid/Ref : xxx.xxx, Late ?-frame(00), Compensated..." messages.

    Is this a symptom which is related to what you guys are discussing? Is there anything I can do? e.g. do you need some logfiles and if yes which ones? Can you point me to some background info?

    I am really not too lazy to read but simply - as said before - lost... can anybody, please, bring me back "on track".

    Thank you,
    René

    P.S. I've spent 3 hours reading this morning... :confused:
     

    Owlsroost

    Retired Team Member
  • Premium Supporter
  • October 28, 2008
    5,540
    5,038
    Cambridge
    Home Country
    United Kingdom United Kingdom
    Re: AW: MediaPortal Audio renderer - better video playback quality

    One thing that can be done is to reduce the drifting (hopefully removing it completely) is to disable the sync correction when bias is 1.0 and refresh rate and source materials fps dont match, since now the EVR presenter is telling to audio renderet that it should adjust the audio resampling a bit, but those are just not working since the major (bias) is not even near the correct speed.

    I'll incorporate a mod for this next time the dshowhelper SVN is updated - just need to test it (I have a sample of a 24Hz-video-in-a-60Hz-wrapper video file from Mironicus already).

    Change it self should be quite minimal.

    It is - one line in fact :)

    Tony
     

    Users who are viewing this thread

    Top Bottom