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Message-ID: <20190131160551.GD2387@tuxbook-pro>
Date: Thu, 31 Jan 2019 08:05:51 -0800
From: Bjorn Andersson <bjorn.andersson@...aro.org>
To: Srinivas Kandagatla <srinivas.kandagatla@...aro.org>
Cc: andy.gross@...aro.org, david.brown@...aro.org,
linux-arm-msm@...r.kernel.org, linux-soc@...r.kernel.org,
linux-kernel@...r.kernel.org, bgoswami@...eaurora.org,
rohitkr@...eaurora.org
Subject: Re: [PATCH] qcom: apr: Make apr callbacks in non-atomic context
On Thu 31 Jan 02:44 PST 2019, Srinivas Kandagatla wrote:
>
>
> On 31/01/2019 01:16, Bjorn Andersson wrote:
> > On Thu 15 Nov 10:49 PST 2018, Srinivas Kandagatla wrote:
> >
> > > APR communication with DSP is not atomic in nature.
> > > Its request-response type. Trying to pretend that these are atomic
> > > and invoking apr client callbacks directly under atomic/irq context has
> > > endless issues with soundcard. It makes more sense to convert these
> > > to nonatomic calls. This also coverts all the dais to be nonatomic.
> > >
> > Hi Srinivas,
> >
> > Sorry for not looking at this before.
> >
> NP, thanks for the review!
>
> > Are you sure that you're meeting the latency requirements of low-latency
> > audio with this change?
>
> Low and Ultra Low Latency audio is not supported in the exiting upstream
> qdsp drivers.
>
Sure, but we want the design to allow for that still, either in future
upstream or by additional downstream code.
> Also it depends on definition of "latency", is the latency referring to
> "filling the data" or "latency between DSP command and response".
>
I'm referring to the latency between the message from the DSP until we
give it a new buffer.
> For former case as long as we have more samples in our ring buffer there
> should be no latency in filling the data.
> For later case it should not really matter as long as former case is taken
> care off.
>
> Low latency audio involves smaller sample sizes and no or minimal
> preprocessing in DSP so am guessing that we should be okay with responses in
> workqueue as long as we have good size ring buffer.
>
Relying on more buffered data will increase the latency of the audio,
preventing you from doing really low-latency things.
Regards,
Bjorn
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