Re: [edk2-devel] VirtIO Sound Driver (GSoC 2021)

["Ethin Probst" <harlydavidsen@gmail.com>]

Also, forgot to add this before sending: yes, speech synthesizers
usually generate the PCM audio on the fly. They can write it to an
output file, but if your just using it in a screen reader, then you
end up streaming it to the audio device. This raises another issue I
was pondering, but I (don't think) we need to handle that quite yet.
The problem involves output generated by the HII, console, etc.: when
we're using a speech synthesizer, it might be configured to speak at a
faster or slower rate depending on the preferences of the user (we
definitely want to let the user control these things because different
people have different preferences on the speed of speech synthesizers:
some can understand it at really fast rates and others can't, for
example). The problem arises when we want to forward output from the
screen (say, the simple text output protocol). Assume that a user is
running the EFI shell as an example, which, if I'm not mistaken, uses
this protocol. The shell calls
EFI_SIMPLE_TEXT_OUTPUT_PROTOCOL.OutputString(). We probably then want
this function to forward the string passed in onto the speech
synthesizer, assuming that accessibility features are enabled (I'm
assuming we'd want to make that a toggle). The problem is that one can
call EFI_SIMPLE_TEXT_OUTPUT_PROTOCOL.OutputString() many times. During
all these calls, text is being sent to the synthesizer, its generating
samples, and forwarding those samples onto the audio output protocol.
The problem is: how do we ensure that these samples don't overlap or
cause other problems (e.g.: interrupt streams that are still being
processed)? As I said, these are things we don't need to necessarily
consider now, but this is a problem we'll need to tackle in the
future.

On 4/16/21, Ethin Probst <harlydavidsen@gmail.com> wrote:
> Yes, three APIs (maybe like this) would work well:
> - Start, Stop: begin playback of a stream
> - SetVolume, GetVolume, Mute, Unmute: control volume of output and enable
> muting
> - CreateStream, ReleaseStream, SetStreamSampleRate: Control sample
> rate of stream (but not sample format since Signed 16-bit PCM is
> enough)
> Marvin, how do you suggest we make the events then? We need some way
> of notifying the caller that the stream has concluded. We could make
> the driver create the event and pass it back to the caller as an
> event, but you'd still have dangling pointers (this is C, after all).
> We could just make a IsPlaying() function and WaitForCompletion()
> function and allow the driver to do the event handling -- would that
> work?
>
> On 4/16/21, Andrew Fish <afish@apple.com> wrote:
>>
>>
>>> On Apr 16, 2021, at 4:34 AM, Leif Lindholm <leif@nuviainc.com> wrote:
>>>
>>> Hi Ethin,
>>>
>>> I think we also want to have a SetMode function, even if we don't get
>>> around to implement proper support for it as part of GSoC (although I
>>> expect at least for virtio, that should be pretty straightforward).
>>>
>>
>> Leif,
>>
>> I’m think if we have an API to load the buffer and a 2nd API to play the
>> buffer an optional 3rd API could configure the streams.
>>
>>> It's quite likely that speech for UI would be stored as 8kHz (or
>>> 20kHz) in some systems, whereas the example for playing a tune in GRUB
>>> would more likely be a 44.1 kHz mp3/wav/ogg/flac.
>>>
>>> For the GSoC project, I think it would be quite reasonable to
>>> pre-generate pure PCM streams for testing rather than decoding
>>> anything on the fly.
>>>
>>> Porting/writing decoders is really a separate task from enabling the
>>> output. I would much rather see USB *and* HDA support able to play pcm
>>> streams before worrying about decoding.
>>>
>>
>> I agree it might turn out it is easier to have the text to speech code
>> just
>> encode a PCM directly.
>>
>> Thanks,
>>
>> Andrew Fish
>>
>>> /
>>>    Leif
>>>
>>> On Fri, Apr 16, 2021 at 00:33:06 -0500, Ethin Probst wrote:
>>>> Thanks for that explanation (I missed Mike's message). Earlier I sent
>>>> a summary of those things that we can agree on: mainly, that we have
>>>> mute, volume control, a load buffer, (maybe) an unload buffer, and a
>>>> start/stop stream function. Now that I fully understand the
>>>> ramifications of this I don't mind settling for a specific format and
>>>> sample rate, and signed 16-bit PCM audio is, I think, the most widely
>>>> used one out there, besides 64-bit floating point samples, which I've
>>>> only seen used in DAWs, and that's something we don't need.
>>>> Are you sure you want the firmware itself to handle the decoding of
>>>> WAV audio? I can make a library class for that, but I'll definitely
>>>> need help with the security aspect.
>>>>
>>>> On 4/16/21, Andrew Fish via groups.io <afish=apple.com@groups.io>
>>>> wrote:
>>>>>
>>>>>
>>>>>> On Apr 15, 2021, at 5:59 PM, Michael Brown <mcb30@ipxe.org> wrote:
>>>>>>
>>>>>> On 16/04/2021 00:42, Ethin Probst wrote:
>>>>>>> Forcing a particular channel mapping, sample rate and sample format
>>>>>>> on
>>>>>>> everyone would complicate application code. From an application
>>>>>>> point
>>>>>>> of view, one would, with that type of protocol, need to do the
>>>>>>> following:
>>>>>>> 1) Load an audio file in any audio file format from any storage
>>>>>>> mechanism.
>>>>>>> 2) Decode the audio file format to extract the samples and audio
>>>>>>> metadata.
>>>>>>> 3) Resample the (now decoded) audio samples and convert (quantize)
>>>>>>> the
>>>>>>> audio samples into signed 16-bit PCM audio.
>>>>>>> 4) forward the samples onto the EFI audio protocol.
>>>>>>
>>>>>> You have made an incorrect assumption that there exists a requirement
>>>>>> to
>>>>>> be able to play audio files in arbitrary formats.  This requirement
>>>>>> does
>>>>>> not exist.
>>>>>>
>>>>>> With a protocol-mandated fixed baseline set of audio parameters
>>>>>> (sample
>>>>>> rate etc), what would happen in practice is that the audio files
>>>>>> would
>>>>>> be
>>>>>> encoded in that format at *build* time, using tools entirely external
>>>>>> to
>>>>>> UEFI.  The application code is then trivially simple: it just does
>>>>>> "load
>>>>>> blob, pass blob to audio protocol".
>>>>>>
>>>>>
>>>>>
>>>>> Ethin,
>>>>>
>>>>> Given the goal is an industry standard we value interoperability more
>>>>> that
>>>>> flexibility.
>>>>>
>>>>> How about another use case. Lets say the Linux OS loader (Grub) wants
>>>>> to
>>>>> have an accessible UI so it decides to sore sound files on the EFI
>>>>> System
>>>>> Partition and use our new fancy UEFI Audio Protocol to add audio to
>>>>> the
>>>>> OS
>>>>> loader GUI. So that version of Grub needs to work on 1,000 of
>>>>> different
>>>>> PCs
>>>>> and a wide range of UEFI Audio driver implementations. It is a much
>>>>> easier
>>>>> world if Wave PCM 16 bit just works every place. You could add a lot
>>>>> of
>>>>> complexity and try to encode the audio on the fly, maybe even in Linux
>>>>> proper but that falls down if you are booting from read only media
>>>>> like
>>>>> a
>>>>> DVD or backup tape (yes people still do that in server land).
>>>>>
>>>>> The other problem with flexibility is you just made the test matrix
>>>>> very
>>>>> large for every driver that needs to get implemented. For something as
>>>>> complex as Intel HDA how you hook up the hardware and what CODECs you
>>>>> use
>>>>> may impact the quality of the playback for a given board. Your EFI is
>>>>> likely
>>>>> going to pick a single encoding at that will get tested all the time
>>>>> if
>>>>> your
>>>>> system has audio, but all 50 other things you support not so much. So
>>>>> that
>>>>> will required testing, and some one with audiophile ears (or an AI
>>>>> program)
>>>>> to test all the combinations. I’m not kidding I get BZs on the quality
>>>>> of
>>>>> the boot bong on our systems.
>>>>>
>>>>>
>>>>>>> typedef struct EFI_SIMPLE_AUDIO_PROTOCOL {
>>>>>>>  EFI_SIMPLE_AUDIO_PROTOCOL_RESET Reset;
>>>>>>>  EFI_SIMPLE_AUDIO_PROTOCOL_START Start;
>>>>>>>  EFI_SIMPLE_AUDIO_PROTOCOL_STOP Stop;
>>>>>>> } EFI_SIMPLE_AUDIO_PROTOCOL;
>>>>>>
>>>>>> This is now starting to look like something that belongs in boot-time
>>>>>> firmware.  :)
>>>>>>
>>>>>
>>>>> I think that got a little too simple I’d go back and look at the
>>>>> example
>>>>> I
>>>>> posted to the thread but add an API to load the buffer, and then play
>>>>> the
>>>>> buffer (that way we can an API in the future to twiddle knobs). That
>>>>> API
>>>>> also implements the async EFI interface. Trust me the 1st thing that
>>>>> is
>>>>> going to happen when we add audio is some one is going to complain in
>>>>> xyz
>>>>> state we should mute audio, or we should honer audio volume and mute
>>>>> settings from setup, or from values set in the OS. Or some one is
>>>>> going
>>>>> to
>>>>> want the volume keys on the keyboard to work in EFI.
>>>>>
>>>>> Also if you need to pick apart the Wave PCM 16 byte file to feed it
>>>>> into
>>>>> the
>>>>> audio hardware that probably means we should have a library that does
>>>>> that
>>>>> work, so other Audio drivers can share that code. Also having a
>>>>> library
>>>>> makes it easier to write a unit test. We need to be security conscious
>>>>> as we
>>>>> need to treat the Audo file as attacker controlled data.
>>>>>
>>>>> Thanks,
>>>>>
>>>>> Andrew Fish
>>>>>
>>>>>> Michael
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Signed,
>>>> Ethin D. Probst
>>>
>>>
>>> 
>>>
>>>
>>
>>
>
>
> --
> Signed,
> Ethin D. Probst
>


-- 
Signed,
Ethin D. Probst