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.
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?
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.
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.
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.
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?
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.