On 23.03.2016 11:04, Tanu Kaskinen wrote:
On Tue, 2016-03-22 at 12:57 +0100, Georg Chini wrote:
On 22.03.2016 12:51, Tanu Kaskinen wrote:
On Tue, 2016-03-22 at 12:33 +0100, Georg Chini wrote:
On 22.03.2016 12:20, Tanu Kaskinen wrote:
On Tue, 2016-03-22 at 10:11 +0100, Georg Chini wrote:
Hi,
when a sink is started, there is some delay before the first sample is
really played.
This delay is a constant part of the sink latency that will be always
present, so the
minimum sink latency cannot go below that start delay.
Would it be acceptable to adjust the latency range for the device after
each unsuspend
to reflect that?
USB devices (those I have access to) for example have a startup delay in
the range of
10ms, but have a latency range that starts at 0.5ms which does not make
a lot of sense
in my opinion.
I don't understand why the startup delay would limit the minimum
latency once the stream is flowing. Imagine a sound card that is
powered by a nuclear power plant. I don't know how long it takes to
start a nuclear power plant, but let's say it takes a couple of days.
Now the sound card startup delay is a couple of days, but there's no
reason that the audio latency has to be a couple of days once the power
plant is running. Where would all that audio be buffered anyway?
Hi Tanu,
you are wrong.
I don't believe you :)
Look at the code of alsa-sink. It never drops samples. The only way to
compensate
for the startup delay would be to drop audio as long as the sink is not
yet playing,
but that is not done. I could try to implement that however and then you
would be
right, but with the current code at least for the alsa-sink the startup
delay will persist.
The sink isn't responsible for dropping samples in any case. The
connection between the start delay and the runtime latency just doesn't
exist at the sink level.
Again, sorry, but you are wrong. The startup delay does not vanish
magically. I can only point you to the code. I have been working with it
for a couple of month now and I know what I am saying. What happens
for USB devices with timer based scheduling is the following:
1) First audio data is written to the card
2) snd_pcm_start() is called
3) More data is written to the card
4) The reported delay of the card goes up exactly by the amount that was
written
5) This repeats a couple of times
6) The card starts playing after 10 -15ms
7) Now there is some difference between the delay and the amount of data
written
This means that the card is forced to buffer audio until it has started
up. This also
means that module-loopback can't do anything about it - see below.
It's module-loopback that introduces that
connection, and if samples are to be dropped, module-loopback is where
that should happen. And indeed, module-loopback should drop samples
when waiting for the sink to start - I've seen a bluetooth situation
where starting the sink took something like 8 seconds to start, and
module-loopback kept buffering all that time, which did not result in a
pleasant phone call experience :)
Well, the version of module-loopback I am using does exactly that, but
the startup delay is not reflected there for alsa-devices. The sink-input
calls pop immediately because the alsa sink code assumes that the sink
is running after calling snd_pcm_start(), but that is just not true. So
module-loopback cannot see the start delay in that case, this is why
the resulting latency is offset by that amount.
By the way, your mails have bad word wrapping. See how it looks:
https://lists.freedesktop.org/archives/pulseaudio-discuss/2016-March/025830.html
Mh, don't know why this happens. Maybe I should not use CR.
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