On 30.01.2018 03:08, NicoHood wrote:
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
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 :/
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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?
Did you check what sample rates are really set for the sink/source
with "pactl list sinks" and "pactl list sources"? What you are setting
on the command line are in fact sample rates for the streams, not
for sink/source itself. There are several places where re-samplers
are used within PA, so that the rate of the stream need not match
the actual sample rate of the device.
Also, if the sink/source is busy (RUNNING or IDLE), PA will refuse to
change the sample rate. So ensure the source is SUSPENDED before
trying to set it to 48kHz.
I don't see a bug yet, apart from the fact that the TOS link apparently
passes on the sample rate of the connected device and ignores the
rate set for the source in PA. But because there is no easy way to
determine the rate of an incoming stream, I cannot see how this
could be fixed in PA.
In general (because of re-sampling) it should work if the source is
set to 48kHz and the sink is at 44.1kHz. In your pacat example
however you are connecting two independent streams, so if the
first stream is running on 48kHz and you tell the second stream
to interpret the data as being 44.1kHz, it cannot work.
Concerning module-loopback and the rate parameter - make sure
the source is SUSPENDED before loading the module. If you have
something like pavucontrol running, the source will never suspend
and will therefore not be able to change rate when a new stream
appears. Please also check with "pactl list sources" while it is
running that the source was really set to 48kHz.
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