[pulseaudio-discuss] alsa sink latency - how to account for startup delay

[Georg Chini]

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.
>
>> I expected a reply like this. The sink is started up at
>> the moment
>> when you write the first data to the buffer, so all following data can
>> only be played
>> when the previous data has been handled. This means, that the startup
>> delay will
>> persist forever unless you are dropping samples (which you would have to
>> do in
>> your example above but which is not done in pulseaudio).
>> Actually you can see that startup delay when you use an oscilloscope because
>> the real latency is off by that amount.
> What is the "real latency" you are talking here? Is it the full end-to-
> end loopback latency or the sink latency? The loopback latency is
> (somewhat) constant, but the sink latency varies all the time, because
> sometimes the sink buffer is full and sometimes it's empty.
>
> -- 
> Tanu
It's end-to-end latency of module-loopback.