MediaPortal Audio renderer - better video playback quality (27 Viewers)

red5goahead

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  • November 24, 2007
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    I have a feeling the C-Media drivers really are bad and if you can get rid of them I would try.
    For DD live are you just using it to output stereo as multi-channel? If so could you not use AC3Filter or FFDShow to encode as Pro-Logic ?

    I tried that driver though I knew no DD Live will be supported. I'm not very enthusiast about Pro-Logic II ;)
     

    jameson_uk

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  • January 27, 2005
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    I tried that driver though I knew no DD Live will be supported. I'm not very enthusiast about Pro-Logic II ;)
    You can encode as DD 5.1 at 640 kbps with AC3Filter / FFDShow so in theory only difference to DDL should be that DDL is done on hardware and the others in software.

    Anyway offtrack so probably best leave it there or start a new thread but was just pointing out that the stock CMedia drivers were doing a fair amount of messing with the audio as it passed through and I am not sure that is a good thing ;)
     

    red5goahead

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  • November 24, 2007
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    I tried that driver though I knew no DD Live will be supported. I'm not very enthusiast about Pro-Logic II ;)
    You can encode as DD 5.1 at 640 kbps with AC3Filter / FFDShow so in theory only difference to DDL should be that DDL is done on hardware and the others in software.
    Anyway offtrack so probably best leave it there or start a new thread but was just pointing out that the stock CMedia drivers were doing a fair amount of messing with the audio as it passed through and I am not sure that is a good thing ;)

    You can't use ac3 encoding with this audio renderer because it have to receive a pcm into its input. when this audio renderer will be able to encode its pcm channel into an ac3 stream through spd/if so your suggested solution will be possible.
     

    tourettes

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  • January 7, 2005
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    New version available for download / testing:

    Major changes:

    • WASAPI is now using event driven mode (pull mode)
    • Added option to specify WASAPI device period (buffer size)
    • Improved lip sync with WASAPI
    • Refresh rate estimator improved
    • Render stats display rationalised
    • Thread pausing logic modified (Owlsroost - please add more info if needed :))
    • Improved handling of DVD looping video menus and video fps > display fps situations

    DevicePeriod setting in more detail:

    Device side buffer size, the samller the more often the audio renderer needs to write to the buffer. 500000 (50ms) by default.

    0 = default from the audio driver
    1 = minimum from the audio driver
    others -> user specified (default 500000 -> 50 ms)


    Binaries are in: https://forum.team-mediaportal.com/637138-post3.html
     

    tourettes

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  • January 7, 2005
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    Seems perfect. No problem with the audio so no more clicks and anoying sounds. No progessive desynch.
    I'll continue tomorrow for longer tests. ;)

    If you get clicks / pops or other odd behavior please try changing the devicePeriod to 50 ms

    -07-2010 01:18:46.280 [ 19c] InitAudioClient using device period from drivers 10 ms

    That default 10 ms is quite demanding for the CPU / audio renderer. Also it wont make any other difference to have it bigger than the latency grows -> pressing stop will play 50 ms extra of the audio buffer - not affecting the A/V sync.
     

    ronilse

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  • July 19, 2005
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    Hi,
    Should this work for me using dvb-t & monogram codecs(did a test but got a very long & not lip-synced audio)?

    Regards
    Roy
     

    tourettes

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  • January 7, 2005
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    Hi,
    Should this work for me using dvb-t & monogram codecs(did a test but got a very long & not lip-synced audio)?

    Not sure, depends on what output format the decoder is using. Log and TS sample would do fine.
     

    red5goahead

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  • November 24, 2007
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    Work also in speedup PAL mode. The info show a refresh rate probably wrong (75,002) different than reclock panel. but everything seem work fine.
    There is 2 files I can't play. it's odd because they are a common h264 video encoded with ac3 audio (2ch). such as other ones
    that have no issue. attach the log. the play start , I hear a click and suddenly MP stop the play
     

    tourettes

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  • January 7, 2005
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    Work also in speedup PAL mode. The info show a refresh rate probably wrong (75,002) different than reclock panel. but everything seem work fine.
    There is 2 files I can't play. it's odd because they are a common h264 video encoded with ac3 audio (2ch). such as other ones
    that have no issue. attach the log. the play start , I hear a click and suddenly MP stop the play

    Stream seems to change (or audio decoder? who knows :D) the used audio format to float which is not supported currently.

    00000003-0000-0010-8000-00aa00389b71 SUBTYPE_IEEE_FLOAT

    Could you upload a sample of such file to:

    Address: ftp.team-mediaportal.com
    Login: tsdump@team-mediaportal.com
    Password: mpuser88

    since it could be a bug in the audio renderer as well (not working ok with the on the fly media type negotiations). Also if you could make sure that ffdshow audio is connected into the audio renderer (and not some other codec). I guess I need to add some logging about the audio decoder into the renderer side (so the information would be available always in the log).
     

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