Re: [PATCH v15 00/16] Add audio support in v4l2 framework

[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]

 




On 5/9/24 06:13, Jaroslav Kysela wrote:
> On 09. 05. 24 12:44, Shengjiu Wang wrote:
>>>> mem2mem is just like the decoder in the compress pipeline. which is
>>>> one of the components in the pipeline.
>>>
>>> I was thinking of loopback with endpoints using compress streams,
>>> without physical endpoint, something like:
>>>
>>> compress playback (to feed data from userspace) -> DSP (processing) ->
>>> compress capture (send data back to userspace)
>>>
>>> Unless I'm missing something, you should be able to process data as fast
>>> as you can feed it and consume it in such case.
>>>
>>
>> Actually in the beginning I tried this,  but it did not work well.
>> ALSA needs time control for playback and capture, playback and capture
>> needs to synchronize.  Usually the playback and capture pipeline is
>> independent in ALSA design,  but in this case, the playback and capture
>> should synchronize, they are not independent.
> 
> The core compress API core no strict timing constraints. You can
> eventually0 have two half-duplex compress devices, if you like to have
> really independent mechanism. If something is missing in API, you can
> extend this API (like to inform the user space that it's a
> producer/consumer processing without any relation to the real time). I
> like this idea.

The compress API was never intended to be used this way. It was meant to
send compressed data to a DSP for rendering, and keep the host processor
in a low-power state while the DSP local buffer was drained. There was
no intent to do a loop back to the host, because that keeps the host in
a high-power state and probably negates the power savings due to a DSP.

The other problem with the loopback is that the compress stuff is
usually a "Front-End" in ASoC/DPCM parlance, and we don't have a good
way to do a loopback between Front-Ends. The entire framework is based
on FEs being connected to BEs.

One problem that I can see for ASRC is that it's not clear when the data
will be completely processed on the "capture" stream when you stop the
"playback" stream. There's a non-zero risk of having a truncated output
or waiting for data that will never be generated.

In other words, it might be possible to reuse/extend the compress API
for a 'coprocessor' approach without any rendering to traditional
interfaces, but it's uncharted territory.



[Index of Archives]     [ALSA User]     [Linux Audio Users]     [Pulse Audio]     [Kernel Archive]     [Asterisk PBX]     [Photo Sharing]     [Linux Sound]     [Video 4 Linux]     [Gimp]     [Yosemite News]

  Powered by Linux