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

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On 09. 08. 24 12:14, Shengjiu Wang wrote:
On Fri, Aug 9, 2024 at 3:25 PM Pierre-Louis Bossart
<pierre-louis.bossart@xxxxxxxxxxxxxxx> wrote:


Then there's the issue of parameters, we chose to only add parameters
for standard encoders/decoders. Post-processing is highly specific and
the parameter definitions varies from one implementation to another -
and usually parameters are handled in an opaque way with binary
controls. This is best handled with a UUID that needs to be known only
to applications and low-level firmware/hardware, the kernel code should
not have to be modified for each and every processing and to add new
parameters. It just does not scale and it's unmaintainable.

At the very least if you really want to use this compress API,
extend it
to use a non-descript "UUID-defined" type and an opaque set of
parameters with this UUID passed in a header.

We don't need to use UUID-defined scheme for simple (A)SRC
implementation. As I noted, the specific runtime controls may use
existing ALSA control API.

"Simple (A)SRC" is an oxymoron. There are multiple ways to define the
performance, and how the drift estimator is handled. There's nothing
simple if you look under the hood. The SOF implementation has for
example those parameters:

uint32_t source_rate;           /**< Define fixed source rate or */
                 /**< use 0 to indicate need to get */
                 /**< the rate from stream */
uint32_t sink_rate;             /**< Define fixed sink rate or */
                 /**< use 0 to indicate need to get */
                 /**< the rate from stream */
uint32_t asynchronous_mode;     /**< synchronous 0, asynchronous 1 */
                 /**< When 1 the ASRC tracks and */
                 /**< compensates for drift. */
uint32_t operation_mode;        /**< push 0, pull 1, In push mode the */
                 /**< ASRC consumes a defined number */
                 /**< of frames at input, with varying */
                 /**< number of frames at output. */
                 /**< In pull mode the ASRC outputs */
                 /**< a defined number of frames while */
                 /**< number of input frames varies. */

They are clearly different from what is suggested above with a 'ratio-
mod'.

I don't think so. The proposed (A)SRC for compress-accel is just one
case for the above configs where the input is known and output is
controlled by the requested rate. The I/O mechanism is abstracted enough
in this case and the driver/hardware/firmware must follow it.

ASRC is usually added when the nominal rates are known but the clock
sources differ and the drift needs to be estimated at run-time and the
coefficients or interpolation modified dynamically

If the ratio is known exactly and there's no clock drift, then it's a
different problem where the filter coefficients are constant.

Same if you have a 'simple EQ'. there are dozens of ways to implement
the functionality with FIR, IIR or a combination of the two, and
multiple bands.

The point is that you have to think upfront about a generic way to pass
parameters. We didn't have to do it for encoders/decoders because we
only catered to well-documented standard solutions only. By choosing to
support PCM processing, a new can of worms is now open.

I repeat: please do not make the mistake of listing all processing with
an enum and a new structure for parameters every time someone needs a
specific transform in their pipeline. We made that mistake with SOF and
had to backtrack rather quickly. The only way to scale is an identifier
that is NOT included in the kernel code but is known to higher and
lower-levels only.

There are two ways - black box (UUID - as you suggested) - or well
defined purpose (abstraction). For your example 'simple EQ', the
parameters should be the band (frequency range) volume values. It's
abstract and the real filters (resp. implementation) used behind may
depend on the hardware/driver capabilities.

Indeed there is a possibility that the parameters are high-level, but
that would require firmware or hardware to be able to generate actual
coefficients from those parameters. That usually requires some advanced
math which isn't necessarily obvious to implement with fixed-point hardware.

 From my view, the really special cases may be handled as black box, but
others like (A)SRC should follow some well-defined abstraction IMHO to
not force user space to handle all special cases.

I am not against the high-level abstractions, e.g. along the lines of
what Android defined:
https://developer.android.com/reference/android/media/audiofx/AudioEffect

That's not sufficient however, we also need to make sure there's an
ability to provide pre-computed coefficients in an opaque manner for
processing that doesn't fit in the well-defined cases. In practice there
are very few 3rd party IP that fits in well-defined cases, everyone has
secret-sauce parameters and options.

Appreciate the discussion.

Let me explain the reason for the change:

Why I use the metadata ioctl is because the ALSA controls are binding
to the sound card.  What I want is the controls can be bound to
snd_compr_stream, because the ASRC compress sound card can
support multi instances ( the ASRC can support multi conversion in
parallel).   The ALSA controls can't be used for this case,  the only
choice in current compress API is metadata ioctl. And metadata
ioctl can be called many times which can meet the ratio modifier
requirement (ratio may be drift on the fly)

This argument is not valid. The controls are bound to the card, but the element identifiers have already iface (interface), device and subdevice numbers. We are using controls for PCM devices for example. The binding is straight.

Just add SNDRV_CTL_ELEM_IFACE_COMPRESS define and specify the compress device number in the 'struct snd_ctl_elem_id'.

					Jaroslav

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





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