<html><head><style type="text/css"></style></head><body><div style="font-family:arial,helvetica,sans-serif;font-size:10pt"><div>Hi, Great news that AC3 passthrough is working. Well done! Are there any issues with AC3 encoding as I am using the A52 plugin from Pulseaudio (without your patch) and required the following "hacks": a. A rate converter in asoundrc: # Rate Converter to 48kHz, needed for Pulseaudio it seems pcm.Filter_RateConvert {     type rate slave {         pcm "Filter_A52Encode" rate 48000     } }b. A timeout=0 in default.pa load-module module-suspend-on-idle timeout=0              # timeout=0 required Even with these "hacks", I found that sometimes I see:I: module.c: Unloading "module-alsa-sink" (index: #4). D: module-always-sink.c: Autoloading null-sink as no other sinks detected. After I see that, the sink disappears and cannot be selected any more in sound properties. 1. Does the ac3-iec958 profile resolve any of these issues? 2. Jack has ac3jack. What would be the pros and cons of doing the AC3 encoding in Pulseaudio instead of in Alsa? Cheers, Mike </div><div style="font-family: arial,helvetica,sans-serif; font-size: 10pt;"> <div style="font-family: arial,helvetica,sans-serif; font-size: 13px;"><hr size="1">From: pl bossart <bossart.nospam@gmail.com> To: General PulseAudio Discussion <pulseaudio-discuss@mail.0pointer.de> Sent: Fri, 9 July, 2010 18:43:55 Subject: Re: [pulseaudio-discuss] {PATCH][RFC] AC3 passthrough support > 1. I was wondering if MP3/DTS/AC3 passthrough is at the discussion point > only or is someone actively looking at developing it? AC3 passthrough works (see patches at start of thread). That said there are still some opens and feedback is still welcome. I guess this will be part of Meego at some point, but since there''s nothing hw specific the discussion takes place in the open. > 2. I think that encoding needs to be considered as part of the design ie. > the possibility of AC3 or MP3 encoding before output to SPDIF. AC3 encoding can be done by selecting the relevant profile (ac3-iec958). The encoding takes place in alsa-lib a52 plugin. I initially removed these profiles but took some flak for it, apparently some people encode their multichannel streams prior to SPDIF transmission. I am afraid MP3 encoding in PulseAudio doesn't make sense, not supported either by SPDIF or HDMI. And for BT use cases, you might as well encode in SBC if you have PCM in the first place. > Perhaps you're already past this point, but there is an Alsa utility iecset > which returns (and can set) whether raw data or PCM is being passed to > SPDIF. So presumably the Pulseaudio passthrough for Alsa just needs to call > the same function that utility does. iecset sets the AES0-3 bytes, but does not handle actual data formatting which is done by the gstreamer ac3iec958 element and by the alsa-lib iec958 plugin. For now I don't even bother to set these bits, they are so unreliable that receivers usually ignore them. > Can someone point me to any design and API docs for Pulseaudio as I'm > interested in learning where and how it connects to Alsa? No real docs. See the code in src/modules/alsa. Only took me a year to figure things out :-( _______________________________________________ pulseaudio-discuss mailing list <a ymailto="mailto:pulseaudio-discuss@mail.0pointer.de" href="mailto:pulseaudio-discuss@mail.0pointer.de">pulseaudio-discuss@mail.0pointer.de</a> <a href="https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss" target="_blank">https://tango.0pointer.de/mailman/listinfo/pulseaudio-discuss</a> </div></div> </div> </body></html>