Re: [pulseaudio-discuss] PulseAudio Network Sink: Respect alternative sample rate and format

2017-08-16 Thread Tanu Kaskinen
On Thu, 2017-08-10 at 20:33 +0200, Maximilian Böhm wrote:
> A bit unlucky but at least now I know that the tunnel sink doesn't
> switch the sample rate – after you initiate a server connection,
> right? So, to be sure, what happens when I start a 48 kHz stream on
> my client?

The tunnel sample rate from the client to the server is determined at
the time you load module-tunnel-sink. The rate is based on what you
configured with the module arguments, or if you didn't configure
anything, then it's the default rate that is configured in the client
machine's daemon.conf.

If the tunnel rate is 44.1 kHz, and you play a 48 kHz stream to the
tunnel sink, resampling will happen on the client machine.

-- 
Tanu

https://www.patreon.com/tanuk
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss


Re: [pulseaudio-discuss] PulseAudio Network Sink: Respect alternative sample rate and format

2017-08-10 Thread Maximilian Böhm
A bit unlucky but at least now I know that the tunnel sink doesn't switch the 
sample rate – after you initiate a server connection, right? So, to be sure, 
what happens when I start a 48 kHz stream on my client?
>  But I’m afraid my client could still be sending a resampled 41,1 kHz signal 
> to the server.
> Also, no idea if the 24-bit format is preserved or cut to 16-bit (I play a 
> lot video tracks in 24-bit).

Your other suggested method via hard coding the PA remote server doesn‘t work 
for my setup because I want to use a native sound card + usb sound card on the 
client too.

Thanks for your highly appreciated response and sorry for my late answer now!


24.07.2017 22:20 Tanu Kaskinen:
> On Sun, 2017-07-23 at 02:59 +0200, Maximilian Böhm wrote:
>> Hello, I’m very happy that I finally got PulseAudio’s awesome network
>> sink feature set up. Works so far, but the AV receiver on the other
>> computer tells me it gets 41,1 kHz whereas my input in the sink from
>> my client computer is 48 kHz (movies).
>> On both computers, the /etc/pulse/daemon.conf is configured like so:
>>
>> default-sample-format = s24le
>> default-sample-rate = 44100
>> alternate-sample-rate = 48000
>>
>> If I set the default sample rate to 48 kHz on the server, the AV
>> receiver gets 48 kHz as well. But I’m afraid my client could still be
>> sending a resampled 41,1 kHz signal to the server. Also, no idea if
>> the 24-bit format is preserved or cut to 16-bit (I play a lot video
>> tracks in 24-bit).
>> My preferred setup would be one in which the server switches from 48
>> kHz to 41,1 kHz dependent on how it gets input, like a desktop. BTW,
>> this works flawlessly over a direct S/PDIF connection from my
>> computers to the AV receiver.
>> I can’t find any documentation on this. Would appreciate your help.
> So you have loaded module-tunnel-sink on the client? The tunnel sink
> doesn't currently support switching the rate depending on what
> applications play to it.
>
> Do you use any local sound cards on the client? If not, then you could
> set "default-server = server_hostname_or_ip" in /etc/pulse/client.conf.
> Then all applications on the client machine will connect directly to
> the server machine. The sample rate that the server will then
> automatically change depending on the applications.
>

___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss


Re: [pulseaudio-discuss] PulseAudio Network Sink: Respect alternative sample rate and format

2017-07-24 Thread Tanu Kaskinen
On Sun, 2017-07-23 at 02:59 +0200, Maximilian Böhm wrote:
> Hello, I’m very happy that I finally got PulseAudio’s awesome network
> sink feature set up. Works so far, but the AV receiver on the other
> computer tells me it gets 41,1 kHz whereas my input in the sink from
> my client computer is 48 kHz (movies).
> On both computers, the /etc/pulse/daemon.conf is configured like so:
> 
> default-sample-format = s24le
> default-sample-rate = 44100
> alternate-sample-rate = 48000
> 
> If I set the default sample rate to 48 kHz on the server, the AV
> receiver gets 48 kHz as well. But I’m afraid my client could still be
> sending a resampled 41,1 kHz signal to the server. Also, no idea if
> the 24-bit format is preserved or cut to 16-bit (I play a lot video
> tracks in 24-bit).
> My preferred setup would be one in which the server switches from 48
> kHz to 41,1 kHz dependent on how it gets input, like a desktop. BTW,
> this works flawlessly over a direct S/PDIF connection from my
> computers to the AV receiver.
> I can’t find any documentation on this. Would appreciate your help.

So you have loaded module-tunnel-sink on the client? The tunnel sink
doesn't currently support switching the rate depending on what
applications play to it.

Do you use any local sound cards on the client? If not, then you could
set "default-server = server_hostname_or_ip" in /etc/pulse/client.conf.
Then all applications on the client machine will connect directly to
the server machine. The sample rate that the server will then
automatically change depending on the applications.

-- 
Tanu

https://www.patreon.com/tanuk
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss


[pulseaudio-discuss] PulseAudio Network Sink: Respect alternative sample rate and format

2017-07-22 Thread Maximilian Böhm
Hello, I’m very happy that I finally got PulseAudio’s awesome network sink 
feature set up. Works so far, but the AV receiver on the other computer tells 
me it gets 41,1 kHz whereas my input in the sink from my client computer is 48 
kHz (movies).
On both computers, the /etc/pulse/daemon.conf is configured like so:

default-sample-format = s24le
default-sample-rate = 44100
alternate-sample-rate = 48000

If I set the default sample rate to 48 kHz on the server, the AV receiver gets 
48 kHz as well. But I’m afraid my client could still be sending a resampled 
41,1 kHz signal to the server. Also, no idea if the 24-bit format is preserved 
or cut to 16-bit (I play a lot video tracks in 24-bit).
My preferred setup would be one in which the server switches from 48 kHz to 
41,1 kHz dependent on how it gets input, like a desktop. BTW, this works 
flawlessly over a direct S/PDIF connection from my computers to the AV receiver.
I can’t find any documentation on this. Would appreciate your help.
___
pulseaudio-discuss mailing list
pulseaudio-discuss@lists.freedesktop.org
https://lists.freedesktop.org/mailman/listinfo/pulseaudio-discuss