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MediaPortal 1
Development
General Development (no feature request here!)
MediaPortal Audio renderer - better video playback quality
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<blockquote data-quote="tourettes" data-source="post: 649215" data-attributes="member: 10858"><p><span style="font-family: 'Arial Black'">New version available.</span> </p><p></p><p>I guess few people have been awaiting this to be released -> <a href="https://forum.team-mediaportal.com/637138-post3.html" target="_blank">https://forum.team-mediaportal.com/637138-post3.html</a></p><p></p><p><span style="font-family: 'Arial Black'">First test relase with following highlights:</span></p><p><span style="font-family: 'Arial Black'"></span></p><ul> <li data-xf-list-type="ul"> Multichannel support</li> <li data-xf-list-type="ul"> On-the-fly AC3 encoding for SPDIF users</li> <li data-xf-list-type="ul"> Support for non 16 bit samples (24 bit integer, 32 bit integer and 32 bit float)</li> <li data-xf-list-type="ul"> Improved the logging for debugging - log more stream events, used audio decoder etc.</li> </ul><p></p><p><span style="font-family: 'Arial Black'">Fixes:</span></p><p></p><ul> <li data-xf-list-type="ul"> When buffer underrun happens don't discard the available data</li> <li data-xf-list-type="ul"> WASAPI shared mode was not working</li> <li data-xf-list-type="ul"> Missing support for the device latency (for A/V sync) - implemented for WASAPI only</li> <li data-xf-list-type="ul"> Improved A/V sync for pause / seeking / start of playback</li> <li data-xf-list-type="ul"> Fixed some wrong logging of dropped audio samples</li> <li data-xf-list-type="ul"> Fixed device period instructions - logging change only</li> <li data-xf-list-type="ul"> Added logging for the default WASAPI audio device when it is used</li> </ul><p></p><p><span style="font-family: 'Arial Black'">Some new settings:</span></p><p></p><p><strong>EnableAC3Encoding</strong></p><p></p><p>Enables AC3 (Dolby Digital on-the-fly encoding for SPDIF). Disabled by default.</p><p></p><p><strong>AC3bitrate</strong></p><p></p><p>Bitrate setting for the AC3 encoding. Default value is 448 kbps, valid values are 192, 224, 256, 320, 384, 448, 512, 576, 640.</p><p></p><p></p><p><span style="font-family: 'Arial Black'">Some notes:</span></p><p></p><p>1) NVIDIA GPU / HDMI drivers wont work with the audio renderer in WASAPI exclusive mode. This seems to be a driver bug.</p><p></p><p>2) AC3 encoding works only WASAPI exclusive mode (shared mode cannot be ever supported - technical limitation. DirectSound might get support some day)</p><p></p><p>3) SPDIF / AC3 supports only 32.0 / 44.1 / 48 KHz sample rates (and the audio HW must support these as well). To overcome this limitation please use ffdshow's internal resampler plugin. Resampling feature will be added to the audio renderer at some later stage. Similar limitation is affecting the WASAPI shared & exclusive mode (but not DirectSound since it has OS provided resampler, to ruin the sound quality <img src="data:image/gif;base64,R0lGODlhAQABAIAAAAAAAP///yH5BAEAAAAALAAAAAABAAEAAAIBRAA7" class="smilie smilie--sprite smilie--sprite8" alt=":D" title="Big Grin :D" loading="lazy" data-shortname=":D" />). If you see <strong>0x88890008</strong> (AUDCLNT_E_UNSUPPORTED_FORMAT) error in the logs, please use ffdshow's resample for the time being to allow non-native HW sample rates to be used.</p></blockquote><p></p>
[QUOTE="tourettes, post: 649215, member: 10858"] [FONT="Arial Black"]New version available.[/FONT] I guess few people have been awaiting this to be released -> [url]https://forum.team-mediaportal.com/637138-post3.html[/url] [FONT="Arial Black"]First test relase with following highlights: [/FONT] [list] [*] 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. [/list] [FONT="Arial Black"]Fixes:[/FONT] [list] [*] 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 [/list] [FONT="Arial Black"]Some new settings:[/FONT] [b]EnableAC3Encoding[/b] Enables AC3 (Dolby Digital on-the-fly encoding for SPDIF). Disabled by default. [b]AC3bitrate[/b] Bitrate setting for the AC3 encoding. Default value is 448 kbps, valid values are 192, 224, 256, 320, 384, 448, 512, 576, 640. [FONT="Arial Black"]Some notes:[/FONT] 1) NVIDIA GPU / HDMI drivers wont work with the audio renderer in WASAPI exclusive mode. This seems to be a driver bug. 2) AC3 encoding works only WASAPI exclusive mode (shared mode cannot be ever supported - technical limitation. DirectSound might get support some day) 3) SPDIF / AC3 supports only 32.0 / 44.1 / 48 KHz sample rates (and the audio HW must support these as well). To overcome this limitation please use ffdshow's internal resampler plugin. Resampling feature will be added to the audio renderer at some later stage. Similar limitation is affecting the WASAPI shared & exclusive mode (but not DirectSound since it has OS provided resampler, to ruin the sound quality :D). If you see [b]0x88890008[/b] (AUDCLNT_E_UNSUPPORTED_FORMAT) error in the logs, please use ffdshow's resample for the time being to allow non-native HW sample rates to be used. [/QUOTE]
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