Hi Pierre,
On 1/6/2023 7:57 AM, Pierre-Louis Bossart wrote:
On 12/23/22 17:31, Wesley Cheng wrote:
Several Qualcomm based chipsets can support USB audio offloading to a
dedicated audio DSP, which can take over issuing transfers to the USB
host controller. The intention is to reduce the load on the main
processors in the SoC, and allow them to be placed into lower power
modes.
It would be nice to clarify what you want to offload
a) audio data transfers for isoc ports
b) control for e.g. volume settings (those go to endpoint 0 IIRC)
c) Both?
Thanks for sharing your experience, and inputs!
It would be the audio related endpoints only, so ISOC and potentially
feedback ep.
That's good, that means there's a common basis for at least two separate
hardware implementations.
This has a lot of implications on the design. ASoC/DPCM is mainly
intended for audio data transfers, control is a separate problem with
configurations handled with register settings or bus-specific commands.
Control would still be handled by the main processor.
Excellent, one more thing in common. Maintainers like this sort of
alignment :-)
There are several parts to this design:
1. Adding ASoC binding layer
2. Create a USB backend for Q6DSP
3. Introduce XHCI interrupter support
4. Create vendor ops for the USB SND driver
Adding ASoC binding layer:
soc-usb: Intention is to treat a USB port similar to a headphone jack.
The port is always present on the device, but cable/pin status can be
enabled/disabled. Expose mechanisms for USB backend ASoC drivers to
communicate with USB SND.
Create a USB backend for Q6DSP:
q6usb: Basic backend driver that will be responsible for maintaining the
resources needed to initiate a playback stream using the Q6DSP. Will
be the entity that checks to make sure the connected USB audio device
supports the requested PCM format. If it does not, the PCM open call
will
fail, and userpsace ALSA can take action accordingly.
Introduce XHCI interrupter support:
XHCI HCD supports multiple interrupters, which allows for events to
be routed
to different event rings. This is determined by "Interrupter Target"
field
specified in Section "6.4.1.1 Normal TRB" of the XHCI specification.
Events in the offloading case will be routed to an event ring that is
assigned
to the audio DSP.
To the best of my knowledge this isn't needed on Intel platforms, but
that's something we would need to double-check.
I think Mathias mentioned that he was looking into adding some XHCI
secondary interrupter support as well. However, it did have some
slightly different requirements compared to what this offloading feature
is trying to do.
I'll first have to split up the XHCI/HCD changes into separate parts
(interrupter specific and offloading specific), and then I'll work with
him to see what can be improved from there.
Create vendor ops for the USB SND driver:
qc_audio_offload: This particular driver has several components
associated
with it:
- QMI stream request handler
- XHCI interrupter and resource management
- audio DSP memory management
When the audio DSP wants to enable a playback stream, the request is
first
received by the ASoC platform sound card. Depending on the selected
route,
ASoC will bring up the individual DAIs in the path. The Q6USB
backend DAI
will send an AFE port start command (with enabling the USB playback
path), and
the audio DSP will handle the request accordingly.
Part of the AFE USB port start handling will have an exchange of control
messages using the QMI protocol. The qc_audio_offload driver will
populate the
buffer information:
- Event ring base address
- EP transfer ring base address
and pass it along to the audio DSP. All endpoint management will now
be handed
over to the DSP, and the main processor is not involved in transfers.
Overall, implementing this feature will still expose separate sound
card and PCM
devices for both the platorm card and USB audio device:
0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D
SM8250-MTP-WCD9380-WSA8810-VA-DMIC
1 [Audio ]: USB-Audio - USB Audio
Generic USB Audio at usb-xhci-hcd.1.auto-1.4,
high speed
This is to ensure that userspace ALSA entities can decide which route
to take
when executing the audio playback. In the above, if card#1 is
selected, then
USB audio data will take the legacy path over the USB PCM drivers,
etc...
You would still need some sort of mutual exclusion to make sure the isoc
endpoints are not used concurrently by the two cards. Relying on
userspace intelligence to enforce that exclusion is not safe IMHO.
Sure, I think we can make the USB card as being used if the offloading
path is currently being enabled. Kernel could return an error to
userspace when this situation happens.
It's problematic for servers such as PipeWire/PulseAudio that open all
possible PCMs to figure out what they support in terms of formats. I am
not sure we can enforce a user-space serialization when discovering
capabilities?
I see...I'm not too familiar yet with all the different implementations
from userspace yet, so that is something I'll need to look up on the
side. Takashi, would you happen to have any inputs with regards to how
flexible PCM device selection can be from the userspace level? If the
offload PCM device can't be supported, can it fallback to another PCM
device?
Intel looked at this sort of offload support a while ago and our
directions were very different - for a variety of reasons USB offload is
enabled on Windows platforms but remains a TODO for Linux. Rather than
having two cards, you could have a single card and addition subdevices
that expose the paths through the DSP. The benefits were that there was
a single set of controls that userspace needed to know about, and volume
settings were the same no matter which path you used (legacy or
DSP-optimized paths). That's consistent with the directions to use 'Deep
Buffer' PCM paths for local playback, it's the same idea of reducing
power consumption with optimized routing.
Volume control would still be done through the legacy path as mentioned
above. For example, if a USB headset w/ a HID interface exposed (for
volume control) was connected, those HID events would be routed to
userspace to adjust volume accordingly on the main processor. (although
you're right about having separate controls still present - one for the
ASoC card and another for USB card)
The two sets of controls implied by the use of two cards is really
problematic IMHO. This adds complexity for userspace to figure out that
the controls are really the same and synchronize/mirror changes.
> The premise of offload is that it should really not get in the way of
user-experience, design constructs that result in delayed starts/stop,
changed volumes or quality differences should be avoided, or
users/distros will disable this optimization.
Makes sense. I think in terms of controls, we know that for an USB
audio device, anything will still be handled through the USB card.
Again, since I'm not too familiar yet with all the userspace
implementations, does it have mechanisms to treat the control and data
interfaces separately?
One card with additional DSP-based PCM devices seems simpler to me in
terms of usage, but it's not without technical challenges either: with
the use of the ASoC topology framework we only know what the DSP
supports when registering a card and probing the ASoC components.
The interaction between USB audio and ASoC would also be at least as
complicated as display audio, in that it needs to work no matter what
the probe order is, and even survive the Linux device/driver model
requirement that there are no timing dependencies in the driver
bind/unbind sequences.
Yes, this was my initial approach as well, but from the technical
perspective it was very very messy, and potentially could have affected
functionality on certain devices if not handled correctly. I think the
difficult part was that the USB SND framework itself is an independent
entity, and it was tough to dissect the portions which created PCM/sound
card devices.
I don't think that was something which would have gone well if
introduced all at once. It would require a lot of discussion before
getting the proper implementation. At least this series introduces the
initial communication between ASoC and USB SND, and maybe as use cases
become clearer we can always improve/build on top of it.
Another point is that there may be cases where the DSP paths are not
available if the DSP memory and MCPS budget is exceeded. In those cases,
the DSP parts needs the ability to notify userspace that the legacy path
should be used.
If we ran into this scenario, the audio DSP AFE port start command would
fail, and this would be propagated to the userspace entity. It could
then potentially re-route to the legacy/non-offload path.
'start' or 'open'? This is a rather important design difference. Usually
we try to make decisions in the .open or .hw_params stage. The 'start'
or 'trigger' are usually not meant to fail due to unavailable resources
in ALSA.
This happens during the .prepare() phase.
Another case to handle is that some USB devices can handle way more data
than DSPs can chew, for example Pro audio boxes that can deal with 8ch
192kHz will typically use the legacy paths. Some also handle specific
formats such as DSD over PCM. So it's quite likely that PCM devices for
card0 and card1 above do NOT expose support for the same formats, or put
differently that only a subset of the USB device capabilities are
handled through the DSP.
Same as the above. We have programmed the USB backend to support the
profiles that the audio DSP can handle. I assume if there was any other
request, the userspace entity would fail the PCM open for that requested
profile.
What's not clear to me is whether there's any matching process between
the DSP capabilities and what the USB device exposes? if the USB device
is already way more complicated that what the ASoC back-end can deal
with, why expose a card?
That's something I thought was done by the ASoC core. I can check that
and see if that's the case. There is a check added in hw_params of our
ASoC component where we do query the USB audio descriptors to ensure
that the PCM format being used is supported by the device. I guess this
is when the DSP capabilities are better than what the headset can
support :).
And last, power optimizations with DSPs typically come from additional
latency helping put the SoC in low-power modes. That's not necessarily
ideal for all usages, e.g. for music recording and mixing I am not
convinced the DSP path would help at all.
That's true. At the same time, this feature is more for power related
benefits, not specifically for performance. (although we haven't seen
any performance related issues w/ this approach on the audio profiles
the DSP supports) I think if its an audio profile that supports a high
sample rate and large number of channels, then the DSP wouldn't be able
to support it anyway, and userspace could still use the legacy path.
This would allow for those high-performance audio devices to not be
affected.
ok, we are aligned as well here. Excellent. With the on-going work to
introduce 'Deep Buffer' capabilities, we'll have a need to tag PCM
devices with a usage or 'modifier', or have this information in
UCM/topology. Logic will have to be introduced in userspace to use the
best routing, I would think this work can be reused for USB cases to
indicate the offload solution is geared to power optimizations.
Great, I like that idea to see if we can help userspace choose the
desired path based on what the overall system is looking for. I wonder
if that would also potentially help with some of the PCM device
selection complexities you brought up as well. If the system just wants
best possible performance then it would just completely ignore the power
optimized (offload) path for any device.
Thanks
Wesley Cheng
