Re: [RFC PATCH 1/6] ALSA: compress: add Sample Rate Converter codec support

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On 09. 08. 24 14:52, Pierre-Louis Bossart wrote:

And metadata
ioctl can be called many times which can meet the ratio modifier
requirement (ratio may be drift on the fly)

Interesting, that's yet another way of handling the drift with userspace
modifying the ratio dynamically. That's different to what I've seen before.

Note that the "timing" is managed by the user space with this scheme.

And compress API uses codec as the unit for capability query and
parameter setting,  So I think need to define "SND_AUDIOCODEC_SRC'
and 'struct snd_dec_src',  for the 'snd_dec_src' just defined output
format and output rate, channels definition just reuse the snd_codec.ch_in.

The capability query is an interesting point as well, it's not clear how
to expose to userspace what this specific implementation can do, while
at the same time *requiring* userpace to update the ratio dynamically.
For something like this to work, userspace needs to have pre-existing
information on how the SRC works.

Yes, it's about abstraction. The user space wants to push data, read data back converted to the target rate and eventually modify the drift using a control managing clocks using own way. We can eventually assume, that if this control does not exist, the drift cannot be controlled. Also, nice thing is that the control has min and max values (range), so driver can specify the drift range, too.

And again, look to "PCM Rate Shift 100000" control implementation in sound/drivers/aloop.c. It would be nice to have the base offset for the shift/drift/pitch value standardized.

					Jaroslav

--
Jaroslav Kysela <perex@xxxxxxxx>
Linux Sound Maintainer; ALSA Project; Red Hat, Inc.





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