On 01/29/2018 07:48 PM, Georg Chini wrote: > On 29.01.2018 19:39, NicoHood wrote: >> On 01/29/2018 07:19 PM, Georg Chini wrote: >>> On 29.01.2018 18:46, Nico wrote: >>>> On 01/29/2018 03:52 PM, NicoHood wrote: >>>>> On 01/29/2018 03:29 PM, Georg Chini wrote: >>>>>> On 29.01.2018 14:44, NicoHood wrote: >>>>>>> On 01/29/2018 01:39 PM, Georg Chini wrote: >>>>>>>> On 29.01.2018 12:52, Nico wrote: >>>>>>>>> Hi, >>>>>>>>> I want to capture the audio stream of my TV with my PCI-E Toslink >>>>>>>>> sound >>>>>>>>> card and play it back on my usb XLR soundcard. The reason why I >>>>>>>>> do that >>>>>>>>> is to use my own music boxes rather than the TV speakers. With >>>>>>>>> toslink + >>>>>>>>> XLR I do not have problems with ground loops etc. >>>>>>>>> >>>>>>>>> I am using the pulseaudio loopback module with its default >>>>>>>>> configuration. The problem is, that the delay between picture and >>>>>>>>> sound >>>>>>>>> is about one second off, and the longer I wait, the more delay it >>>>>>>>> gets >>>>>>>>> (30 seconds and more). It is no usable like this. >>>>>>>>> >>>>>>>>> I tried to play with the latencies of source, sink and the tv >>>>>>>>> delay >>>>>>>>> itself without sucess. I also tried streaming with pacat >>>>>>>>> directly as >>>>>>>>> described here: >>>>>>>>> https://thelinuxexperiment.com/fix-pulseaudio-loopback-delay/ >>>>>>>>> >>>>>>>>> I also tried to change different parameters of the loopback >>>>>>>>> module or >>>>>>>>> the sources/sinks, but that did not help. I never changed any >>>>>>>>> global >>>>>>>>> pulseaudio config to avoid larger configuration issues. The CPU >>>>>>>>> usage of >>>>>>>>> pulseaudio is at 3% with the loopback module >>>>>>>>> >>>>>>>>> Can anyone help me to get rid of this lag? >>>>>>>>> >>>>>>>> Hi Nico, >>>>>>>> >>>>>>>> which version of PA are you using? Can you provide logs? >>>>>>>> >>>>>>>> Regards >>>>>>>> Â Â Â Â Â Â Â Â Â Â Â Â Â Â Â Georg >>>>>>>> >>>>>>> Hi Georg, >>>>>>> oh sure I completely forgot: >>>>>>> >>>>>>> pulseaudio 11.1-1 (Arch Linux) >>>>>>> uname -a: Linux zebes 4.14.15-1-ARCH #1 SMP PREEMPT Tue Jan 23 >>>>>>> 21:49:25 >>>>>>> UTC 2018 x86_64 GNU/Linux >>>>>>> >>>>>>> Here is a logfile: >>>>>>> LANG=C pulseaudio -vvvv --log-time=1 > ~/pulseverbose.log 2>&1 >>>>>>> https://gist.github.com/NicoHood/85976f426e1621e599253ee1a95230dd >>>>>>> >>>>>>> Regards >>>>>>> Nico >>>>>> This is weird. It looks like the source sample rate is so much higher >>>>>> than the sink rate that module-loopback can't adapt. No idea why >>>>>> this happens. Does it work with another input? >>>>>> >>>>>> >>>>> I've tested it also with the builtin front microphone input (2nd >>>>> revision on gist) and with the PCIE Analog line in (3rd gist >>>>> revision). >>>>> You can view the changes here: >>>>> https://gist.github.com/NicoHood/85976f426e1621e599253ee1a95230dd/revisions >>>>> >>>>> >>>>> >>>>> The problem only occurs with the digital, optical TOSLINK input >>>>> from my >>>>> Samsung TV. For some other reason the sound distortion for the analog >>>>> input is now (temporary) gone. However I still want to get that >>>>> TOSLINK >>>>> running :/ >>>>> _______________________________________________ >>>>> pulseaudio-discuss mailing list >>>>> pulseaudio-discuss at lists.freedesktop.org >>>>> https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss >>>>> >>>> As an additional test I send the audio to the TOSLINK output of the >>>> soundcard, back to its input and setup loopback on the input again (to >>>> the USB soundcard). With this method the sound has no lag. The >>>> difference is that the toslink signal comes from the soundcard itself >>>> rather than the TV. I have some smaller noises/distortion however >>>> sometimes. Here is the log: >>>> https://gist.github.com/NicoHood/6ef237649d97eaa69d9a78dd91eff34a >>>> >>> Somehow this still sounds like a sample rate mismatch. (A wild >>> guess, I do not know how the signal chain works for a TOS-link) >>> Have you tried setting your card to 48kHz? You can change >>> default-sample-rate and alternate-sample-rate in daemon.conf >>> both to 48000 and restart pulse to enforce this. >>> >>> >> With 48kHz The sound is working properly now. The problem I got ~~now~~ >> is that there is sound distortion in the recording. -> A reboot fixed >> that. >> >> Is there any negative effect when using 48kHz as default sample rate? Is >> there a way to configure only the loopback module with this special >> sample rate? (Sorry, I have no idea how audio works internally). And if >> there is no other way than configuring the default sample rate, how can >> is set this as user (without root) if possible? >> >> Thanks a lot so far :) >> > If the sound card supports it, PA will always use either the default > or the alternative sample rate, whatever fits the current stream > better, unless there is already a stream running on the card. > > So it should be enough to run module-loopback with rate=48000. > Don't forget to change the values in daemon.conf back to their > original values before you try it. > > Setting the rate for the loopback module does not give a positive result. Setting the sample rate with pacat for the SINK works fine though. This is the command I used: pacat -r --latency-msec=100 -d alsa_input.pci-0000_01_00.0.iec958-stereo | pacat -p --latency-msec=100 -d alsa_output.usb-M-AUDIO_M-Track_Hub-00.analog-stereo --rate=48000 Is this a general bug? Is there a way to change the sample rate for the sink itself within the loopback module? I am still wondering why this happens, as I can capture any other input without changing the sink sample rate. If i set the source to 48000 and the sink to 44100 it does not work. I have no idea why. Any ideas?
