[pulseaudio-discuss] Toslink capture looback latency problem

Georg Chini georg at chini.tk
Mon Jan 29 18:48:17 UTC 2018


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
>>>>>>>                 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.



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