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


Retired Team Member
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  • January 7, 2005
    Beta warning

    Do not use if you aren't willing to take part into the beta testing!

    What is the target audience for audio renderer?

    MediaPortal Audio renderer is currently targeted for people who are after better video playback quality (strange isn't it? :D). Currently HIFI people wont get any benefits from the audio renderer (lets see if that changes in the future).

    ...also this is only alpha quality on both stability and feature level so the current "release" is not intented to be used for anything else than for reporting the errors to make future targets (MP2 mainly) better.

    Benefits of audio renderer

    Audio renderer tries to combat few video playback related issues that almost all (all without Reclock) Windows based (other OSes most likely fall into this category as well) media players are affected:

    Audio / Video clock drifting

    One of the basic issues that almost every sine PC has is the fact that audio and video clocks are based on the different HW crystals / time sources. Normaly this can be only over come by either dropping or inserting silence into the audio stream which causes pops & clics in audio stream or by dropping or repeating video frames which result juddering / non-smooth video playback.

    MediaPortal Audio renderer solves this (like Reclock as well) by adjusting the reference clock a bit and resampling the audio to match the result. This will slightly degenerate audio quality when ever the correction is done.

    Refresh rate and fps mismatch

    Like Reclock, MediaPortal Audio renderer is able to speed up / down the playback speed of the media to make the media fps to match the dispay refresh rate or it's whole number multiplier.

    EVR presenter's vsync position

    EVR presenter is able to "steer" the audio renderer in such way that the video frame rendering is done on optimal vsync position. This is able to correct the random EVR mixer vsync position that is "selected" on streaming start (playback start, skip or resume from pause).

    Limitations of audio renderer

    Currently there are following limitations for MediaPortal Audio renderer

    • Bitstreaming (no point since resampling destroys the whole idea of bitstreaming) - includes all formats
    • Windows 7 (Vista most likely as well) audio decoder is not supporting AC3 with 3rd party applications (Dolby lisencing restirction - Microsoft MPEG-1/DD/AAC Audio Decoder (Windows))
    • Volume control is not working with WASAPI exclusive mode since it bypasses the OS audio mixer.

    Future possibitilies of audio renderer

    • Bitstreaming (this will cause lot of the main, video playback related features not to work - so this is not on high priority)

    Already done:
    • Not possible the playback speed adjust window / bias control
    • Implement WASAPI event driven mode (polling mode wont be available after that)
    • Support for multi channel audio
    • Support for non-16 bit samples
    • AC3 re-encoding so that SPDIF users can have use on the resampling functionality (without this only HDMI LPCM or analog would allow multi channel audio formats when multi channel support is implemented)
    • Audio HW based reference clock
    • Implement a "learning" bias that will reduce the amount of audio adjustments done for non 1:1, 1:2, 1:3... rates

    How to install

    • backup current dshowhelper.dll
    • copy the attached dshowhelper.dll over the old one
    • copy to some folder
    • use "regsvr32 mpaudiorenderer" to register the audio renderer
    • set "MediaPortal audio renderer" to be used in DVD / TV / My Videos as audio renderer

    Debug settings for audio renderer

    Following registry settings can be used to tweak / debug / troubleshoot the audio renderer's functionality. They are located in HKEY_CURRENT_USER\Software\Team MediaPortal\Audio Renderer


    Enables all the audio renderer's special features to work (should be only disabled to check if some issue is also reporducible with it disabled). Enabled by default.


    Forces DirectSound to be used in Vista / Windows 7 instead of the WASAPI. Disabled by default.


    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)


    Uses WASAPI exclusive mode. Enabled by default.


    Uses WASAPI event driven mode. Enabled by default (causes issues on some HW / drivers where no proper support is available.)


    Audio end point device (where the audio is played). Check AudioRenderer.log for possible devices. If left empty it will use OS default which should be ok for most of the time.


    Toggles between the audio HW based reference clock and system time based reference clock. Audio HW based reference clock is the default value (1).


    Toggles between v-sync adjustments on / off. By default this is on. Use only for debugging purposes (like sync drifting).


    For debugging only. When enabled, all audio incoming sample timestamps (and current stream time) is logged into the audiorenderer.log. Disabled by default.


    Enables AC3 (Dolby Digital on-the-fly encoding for SPDIF). Disabled by default.


    Bitrate setting for the AC3 encoding. Default value is 448 kbps, valid values are 192, 224, 256, 320, 384, 448, 512, 576, 640.


    Maximum allowed speed that the playback can reach, this value is divided with 10000 to get the real value. Default value is 11000 (1.1x) - 10% speed up at maximum.


    Minimum allowed speed for the playback, this value is divided with 10000 to get the real value. Default value is 9000 (0.9x) - 10% speed down at maximum.


    Delay audio for xxx ms, can be used to compensate delay introduced by display device (postprocessing in LCD for example). By default this is 0 ms. Amount is given in ms.


    Override the channel mask for 5.1 stream types. Use this to fix incompatibilities with the speak configuration between driver and the audio decoder. Disabled / zero by default. For ASUS Xonar use "1551".


    Override the channel mask for 7.1 stream types. Use this to fix incompatibilities with the speak configuration between driver and the audio decoder. Disabled / zero by default.

    How to configure audio codecs

    Since MediaPortal audio renderer is currently really picky about the input audio stream format that it accepts it is recommended to use ffdshow's audio decoder (which is a really good decoder in any case). Following setting should be used:

    • All passthru settings must be disabled
    • AC3 encode should be disabled

    How to report a bug

    • Try to provide as clean logs as possible (for example if live tv causes issues, just after the last channel change if possible)
    • Provide information about the used sample media file (even better if you can provide a small piece of that file itself, but take care that you aren't violating copyright)
    • Provide system information about the audio chip & drives. Same for GPU
    • Provide AudioRenderer.log
    • Provide Evr.log
    • Provide MediaPortal.log
    • Check with GraphEdit / GraphStudio that the audio renderer is really connected to the audio decoder in graph

    Source codes

    EVR presenter and audio renderer source codes can be found from

    public - Revision 26095: /branches/Accurate_sync

    People who made this possible

    ar-jar (Arto Järvinen - vertical sync). The main idea for searching for the better video playback quality (mainly the A/V clock drift issue we are plagued) did come from Arto's work on the GothPlayer / MPC-HC side.

    Arion_p - For helping the audio renderer itself (and volunteering to continue with the multi channel implementation :))

    Olwsroost - He was the one who did continue the EVR presenter scheduling and frame presentation improvements.

    albain - ffdshow developer who has provided the MPC-HC project the basic audio renderer implementation that the MediaPortal Audio renderer is based on.

    The attached picture has been taken on approx 105 minutes of time when a Blu-ray sourced M2TS file is being played on non-matching 50 Hz monitor.



    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    Open issues:
    • ATI GPU with HDMI connection fails to accept 24 bit streams. This seems to be a limitation of HW or drivers (in the future resampling feature will fix this if no driver side support is done by manufacturer).
    • Missing support for the device latency (for A/V sync) - DirectShow only (implemented for WASAPI)
    • AC3 encoding is not supported by DirectSound mode

    Resolved issues:
    • ASUS Xonar cards have problems with WASAPI exclusive & even driven mode - issue can be reproduced with the MS play-exclusive.exe as well - a workaround has been added.
    • NVIDIA GPU with HDMI connection fails to play anything in exclusive WASAPI mode (AUDCLNT_STREAMFLAGS_EVENTCALLBACK is not supported?) ----> resolved in Nvidia HDMI driver revision
    • DirectSound creates lot of "receiveSamples: Output sample queue was empty!"
    • WASAPI shared mode is not working
    • Missing support for the device latency (for A/V sync) - DirectShow only (implemented for WASAPI)
    • A/V sync can get apart when there is corrupted / dropped data in the stream
    • Looping noise in some cases when no audio data is available to be rendered
    • Audio render will not render the few last seconds of the audio (depending on the video container / muxing parameters used) - fixed in SVN, will be included in the next version
    • FLAC audio in MKV could cause buffer underruns when time stretching is enabled
    • DVD menus can trigger main movie to nave no sound and then hang (WASAPI only)


    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    version 1 - based on SVN revision 26064

    Initial version.

    version 2 - based on SVN revision 26095

    First version with WASAPI event driven mode in use

    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

    version 3 - based on SVN revision 26100

    Minor patch release.

    • Fixed A/V sync going of when stream contains corrupted / dropped data (live tv / recorded tv most likely)
    • Fixed audio getting into looped state when no new data is available (for example signal lost in TV for a brief moment) - now only silence will get rendered in such cases
    • Improved handling of DVD looping video menus (Astronaut Farmer's main menu is now working 100% ok)

    version 4 - based on SVN revision 26101

    Minor patch release.

    • Added registry setting for sample time logging
    • Fixed dropped data detection after seeking

    version 5 - based on SVN revision 26107

    Minor patch release.

    • Audio render will not render the few last seconds of the audio (depending on the video container / muxing parameters used) - fixed in SVN, will be included in the next version
    • FLAC audio in MKV could cause buffer underruns when time stretching is enabled
    • Disabled audio clock fine tuning when refresh rate and source material doesn't match 1:1, 1:2 or 1:3 (even after possible speed up / slow down)

    version 6 - based on SVN revision 26202

    First test relase with following highlights:

    • Multichannel support
    • On-the-fly AC3 encoding for SPDIF users
    • Support for non 16 bit samples (24 bit integer, 32 bit integer and 32 bit float)
    • Improved the logging for debugging - log more stream events, used audio decoder etc.


    • When buffer underrun happens don't discard the available data
    • WASAPI shared mode was not working
    • Missing support for the device latency (for A/V sync) - implemented for WASAPI only
    • Improved A/V sync for pause / seeking / start of playback
    • Fixed some wrong logging of dropped audio samples
    • Fixed device period instructions - logging change only
    • Added logging for the default WASAPI audio device when it is used

    version 7 - based on SVN revision 26225

    Minor patch release.


    • DVD menus can trigger main movie to nave no sound and then hang (WASAPI only)
    • Removed the audio buffer flush when pausing audio, this allows audio to continue sooner after the resume is done. There is still small glitch - amount is equal to the audio device's latency
    • Introduced preroll samples. This will remove the need for dropping the first audio samples to match the device latency
    • Reduced amount of the audio samples that are dropped when discontinuity in the stream is detected (valid samples from "past" are renderer)

    version 8 - based on SVN revision 26250

    Minor patch release.


    • Aligned WASAPI buffer to match AC3 frame lenght when AC3 encoding is enabled - should reduce noise when zapping / seeking
    • Fixed Last WASAPI buffer before seeking / stop / pause getting cut
    • Fixed a memory / resource leak
    • Added logging for sample allocator failures (during creation)
    • Fixed audio adjustment to be thread safe - earlier it was possible that audio channels would have been out of phase in some cases when speed adjustment was done
    • "Fixed wrong caching of alphablend" changes from mr_hipp incorporated. Render stats re-positioned so they're not covered by the timeline bar.

    version 9 - based on SVN revision 26274

    Test relase with following highlight:

    • Added support for WASAPI polling based buffer filling (should help with HW / drivers that are broken - NVIDIA, Xonar)


    • MMCSS thread boost was not done for the rendering thread
    • Added OS version detection - WASAPI is not even tried to be enabled on Windows XP
    • Few rare crashes - not sure if those have been present in earlier releases (SVN only most likely)
    • Small REW/FFDW improvements in EVR presenter

    version 10 - based on SVN revision 26297

    Test relase with following highlight:

    • Added channel mask override settings for 5.1 and 7.1 stream types. This is required for example ASUS Xonar


    • Try WAVEFORMATEXin format negotiation if the WAVEFORMATEXTENSIBLE is reported to be unsupported by the driver.

    version 11 - based on SVN revision 26308

    Minor fix release.

    NOTE: you need to reset the 5.1 and 7.1 channel mask overrides - setting name has been changed in registry

    channelMaskOverride_5_1 to ChannelMaskOverride_5_1
    channelMaskOverride_7_1 to ChannelMaskOverride_7_1

    • Fixed a deadlock with audio stream changes
    • Fixed A/V sync with stream changes
    • Fixed a possible crash with the AC3 encoding

    version 12 - based on SVN revision 26348

    Minor fix release.

    • Refactored A/V sync / sample scheduling - hopefully this will sort out few A/V sync related issues, for example the one that Xonar has with big buffers
    • Fixed sample handling with the time strecthing disabled - madFlac was not working at all. ffdshow could have lots of dropped audio data.
    • Fixed format negotiation when EnableTimestretching was set to disabled
    • Added logging for device reported latency.

    version 13 - based on SVN revision 26419

    Initial version of the audio HW based reference clock. No other changes.

    version 14 - based on SVN revision 26527

    Minor fix release.

    • Fixed a crash with the OnlineVideos plugin
    • Fixed a deadlock on stream automatic type changes
    • Minor adjustment to rendering stats layout
    • Added workaround for the MS Video decoder reporting wrong fps values and therefore causing incorrect bias calculation.
    • Improving bias calculation accuracy
    • Added a setting for toggling the sync adjustment on/off for debugging. When disabled the EVR presenter is not able to do the minor v-sync position adjustments.
    • Improving A/V sync - drifting is tried to be minimized (drift handling needs to be refactored completely to fix it completely)

    version 15 - based on SVN revision 26591

    • Completely refactored the A/V sync drift handling - thanks davidf
    • Fixed a memory leak

    version 16 - based on SVN revision 26614

    • Reduced duration of audio resampling that is required for 1:1, 1:2 and 1:3 material - thanks davidf
    • Changed collected drift data to be reset only on graph ::run. Previously this was done when ever audio clock was "skipping" backwards / having discontinuity which could cause the reset to be done in incorrect times with some drivers (Xonar at least)
    • Small tweaks to the rendering stats

    version 17 - based on SVN revision 26623

    • Added audio delay setting in the registry
    • Fixed A/V sync drifting when AC3 encoding is enabled

    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

    version 19 - based on SVN revision 26757

    • Fixed WASAPIUseEventMode setting registry key was not created correctly if all settings were removed.
    • Added a workaround for the Xonar audio HW drivers (audio clock produces incorrect values)
    • Fixed a crash when no WASAPI devices are available

    version 20 - based on SVN revision 26834

    • Fixed stream duration calculation - caused issues at least with WMA3 audio codec

    version 21 - based on SVN revision 27950 (added by Owlsroost)

    version 21a (added by Owlsroost)

    • Rebuild of version 21 to remove VC90 dependency - no functional or code changes.



    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    WASAPI support test tool

    Here's a small command line test utility for testing out the WASAPI driver / HW support. It will test all stream configurations and will output the results into the command promt

    Please note that some audio drivers might cause pop / boom etc noises to happen during the testing (even when we are rendering only empty samples with a silent flag turned on!). So make sure that the volume level is not too high.

    How to use

    At first make sure the command promt history size is big enough (few thousand lines should be enough).

    To list all available WASAPI devices use "WASAPI_test.exe --list-devices"

    To test all supported modes use "WASAPI_test.exe --device "device same from the previous command"

    To test all supported modes use and to gain errors for the failures "WASAPI_test.exe --device "device same from the previous command" --verbose

    How to read the results

      192000 16  2   768000  4      1  0         - Format works ok
      192000 16  2   768000  4  65534 22 16    3 - Format works ok
      192000 24  2  1152000  6      1  0         - not supported
      192000 24  2  1152000  6  65534 22 24    3 - not supported
      192000 32  2  1536000  8  65534 22 24    3 - Format works ok
    Here's the 1st line data "analysed":
    Sample rate 192 KHz, 16 bits, 2 channels, 768000 average bytes per second, 4 bytes are needed for one frame, WAVE_FORMAT_PCM

    Here's the 2nd line data "analysed":
    Sample rate 192 KHz, 16 bits, 2 channels, 768000 average bytes per second, 4 bytes are needed for one frame, WAVEFORMATEXTENSIBLE, 22 bytes data, channel mask 3

    Both formats were accepted by the driver / HW.

    Version history

    Version 2

    • Added support for alternative speaker mask testing for 5.1 and 7.1 formats

    Version 3

    • Introduced --verbose command line swich (off by default) - only valid formats get logged by default



    MP Donator
  • Premium Supporter
  • November 24, 2007
    Italy, North West
    Italy Italy
    Right now no Fps speedup Pal features is active. Is it correct?

    edit: Don't forget about time stretching features for the future release. Reclock is based on an opensource library (SoundTouch Sound Processing Library)

    Nome completo : D:\Media\****N.mkv
    Formato : Matroska
    Dimensione : 1,09 GiB
    Durata : 43min
    BitRate totale : 3 606 Kbps
    Data di codifica : UTC 2010-04-29 21:19:15
    Creato con : mkvmerge v3.0.0 ('Hang up your Hang-Ups') built on Apr 19 2010 11:58:49
    Compressore : libebml v0.7.7 + libmatroska v0.8.1

    ID : 1
    Formato : AVC
    Formato/Informazioni : Advanced Video Codec
    Profilo del formato : High@L4.1
    Impostazioni del formato, CABAC : Si
    Impostazioni del formato, ReFrames : 6 fotogrammi
    Modalità di muxing : Container profile=Unknown@4.1
    Codec ID : V_MPEG4/ISO/AVC
    Durata : 43min
    BitRate : 3 220 Kbps
    Larghezza : 1 280 pixel
    Altezza : 720 pixel
    AspectRatio : 16:9
    FrameRate : 23,976 fps
    ColorSpace : YUV
    ChromaSubsampling : 4:2:0
    BitDepth/String : 8 bits
    Tipo di scansione : Progressivo
    Bit/(Pixel*Frame) : 0.146
    Dimensione della traccia : 977 Mb (87%)
    Compressore : x264 core 80 r1376 3feaec2
    Impostazioni compressione : cabac=1 / ref=6 / deblock=1:0:0 / analyse=0x3:0x113 / me=umh / subme=9 / psy=1 / psy_rd=1.0:0.0 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=0 / chroma_qp_offset=-2 / threads=6 / sliced_threads=0 / nr=0 / decimate=1 / mbaff=0 / constrained_intra=0 / bframes=3 / b_pyramid=0 / b_adapt=1 / b_bias=0 / direct=1 / wpredb=1 / wpredp=2 / keyint=250 / keyint_min=25 / scenecut=40 / rc_lookahead=40 / rc=2pass / mbtree=1 / bitrate=3220 / ratetol=1.0 / qcomp=0.60 / qpmin=10 / qpmax=51 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / ip_ratio=1.40 / aq=1:1.00
    Lingua : Inglese

    ID : 2
    Formato : AC-3
    Formato/Informazioni : Audio Coding 3
    Format_Settings_ModeExtension : CM (complete main)
    Codec ID : A_AC3
    Durata : 43min
    Modalità : Costante
    BitRate : 384 Kbps
    Canali : 6 canali
    Posizione dei canali : Front: L C R, Side: L R, LFE
    SamplingRate : 48,0 KHz
    Dimensione della traccia : 119 Mb (11%)


    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    Right now no Fps speedup Pal features is active. Is it correct?
    It should be active. Currently it is having hard coded value to do it always when refresh rate and FPS (or it's multiplier) are less than 6% away each other (for example 23.976 FPS is always speed up to match 25/50Hz). I'll add that information to the 1st post.

    edit: Don't forget about time stretching features for the future release. Reclock is based on an opensource library (SoundTouch Sound Processing Library)
    Timestretching is already done by the same SoundTouch library.


    Retired Team Member
  • Premium Supporter
  • January 7, 2005
    I've tried also with 50 HZ plasma refresh rate tahn 75 Hz. but the speedup seems not active. ;)
    Please post the logs. ! stats are showing the speed up (bias non 1.0 value) but the graph indeed shows that the speed up is not done.


    Retired Team Member
  • Premium Supporter
  • October 17, 2007
    Germany Germany
    AW: MediaPortal Audio renderer - better video playback quality

    did some quick tests. screenshot/logs attached and video details out of MPC-HC

    Complete name : *.avi
    Format : AVI
    Format/Info : Audio Video Interleave
    File size : 682 MiB
    Duration : 9mn 56s
    Overall bit rate : 9 587 Kbps
    Writing application : MEncoder 2:1.0~rc2-0ubuntu13
    Writing library : MPlayer

    ID : 0
    Format : MPEG-4 Visual
    Codec ID : MP42
    Codec ID/Info : Microsoft MPEG-4 v2 (pre-standard)
    Codec ID/Hint : Microsoft
    Duration : 9mn 56s
    Bit rate : 9 328 Kbps
    Width : 1 920 pixels
    Height : 1 080 pixels
    Display aspect ratio : 16:9
    Frame rate : 24.000 fps
    Bits/(Pixel*Frame) : 0.187
    Stream size : 663 MiB (97%)

    ID : 1
    Format : MPEG Audio
    Format version : Version 1
    Format profile : Layer 3
    Mode : Joint stereo
    Codec ID : 55
    Codec ID/Hint : MP3
    Duration : 9mn 56s
    Bit rate mode : Variable
    Bit rate : 246 Kbps
    Channel(s) : 2 channels
    Sampling rate : 48.0 KHz
    Stream size : 17.5 MiB (3%)
    Alignment : Aligned on interleaves
    Interleave, duration : 24 ms (0.58 video frame)
    Interleave, preload duration : 471 ms

    frame drop was while loading stats! (u know... ION and stats don't really like each other :p )


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