[alsa-devel] [PATCH 0/3] ASoC: Enable a new IC master mode: bcm2835<=>IC<=>cs42xx8
hias at horus.com
Mon Feb 27 01:14:59 PST 2017
On Sun, Feb 26, 2017 at 09:41:11PM +0100, Emmanuel Fusté wrote:
> Le 26/02/2017 à 15:49, Matthias Reichl a écrit :
> >On Sun, Feb 26, 2017 at 09:13:09AM +1100, Matt Flax wrote:
> >>On 26/02/17 00:39, Matthias Reichl wrote:
> >>>On Sat, Feb 25, 2017 at 04:03:11PM +1100, Matt Flax wrote:
> >>>>This patch set lets the ASoC system specify that an IC between the SoC and codec
> >>>>is master. This is intended to put both the SoC and Codec into slave modes.
> >>>>By default un-patched SoC and Codec drivers will return -EINVAL if they aren't
> >>>>enabled and tested for this mode.
> >>>>soc-dia.h has the new SND_SOC_DAIFMT_IBM_IFM definition set as :
> >>>>#define SND_SOC_DAIFMT_IBM_IFM (5 << 12) /* IC clk & FRM master */
> >>>>The cs42xx8 codec driver is enabled for this mode and so too is the BCM2835
> >>>>SoC driver. This forms a chain : bcm2835<=>IC<=>cs42xx8
> >>>>where the IC is bit and frame master.
> >>>Model your IC as a codec. No need to add patches to random drivers
> >>>and add a flag with the rather meaningless semantics "someone else is
> >>>automagically setting up clocks for me".
> >>My last patch, used the two codec approach, however it was pointed out that
> >>bcm2835 was run in DSP mode with a codec master (rather then IC master) and
> >>the patch doesn't work. Which is clearly true and a problem, it can only
> >>work with an
> >>intermediate non-codec master.
> >>I think you summed it up well with your statement :
> >>On 25/02/17 Matthias Reichl wrote:
> >>If the clock timing adheres to DSP mode A timing and you add code
> >>to the the CPU DAI driver so it can operate in DSP mode A then
> >>that should also work. If not, it's broken.
> >Your bcm2835 patch doesn't configure the bcm2835 to DSP mode A,
> >it's still setup for I2S (slave) mode. You are just adding code
> >to pretend it's running in DSP mode A. Don't do that, it's wrong.
> >>This patch set fixes the problem of a daisy chain of three possible masters
> >>(CPU <=> IC <=> codec) where only the IC can be master. In fact, when retro
> >>fitting DSP mode to old silicon, the CPU can specify which of the three can
> >>be masters
> >>and there is no chance that someone can fire the system up with the wrong
> >>(which we know produces bit offset and random channel swapping when a codec
> >Please follow the advice I gave you about 3 weeks ago and model your
> >setup properly.
> >| So you have bcm2835 I2S <-> FPGA <-> codec - IOW a standard codec<->codec
> >| link.
> >| What you seem to be missing is just a method to transfer your 8-channel
> >| data via a 2-channel link - userspace want's to see an 8-channel PCM,
> >| but the hardware link (bcm2835-i2s) is only 2-channel.
> >| And that's where IMO as userspace plugin looks like a very good solution.
> >| It's basically the counterpart of your FPGA and contains the code that's
> >| neccessary to encapsulate/pack/whatever the 8-channel data into a 2-channel
> >| stream so it can then be unpacked to 8-channel by the FPGA.
> >| If you go this route your hardware and machine driver will work with
> >| other I2S codecs as well, and IMO that's a far better solution than
> >| adding very ugly hacks to a single I2S driver.
> >If you add an active hardware component (your "IC"/FPGA) you also
> >have to model that in software.
> >If that component is acting as a clock master it probably has some
> >method to setup clocks. Even if you don't have that, eg if you
> >are running at some fixed rate you'll have to store that information
> >The place to do that is in a codec driver. In your setup it'll look
> >like this:
> >That "IC" codec has 2 DAIs and operates as a clock master on both.
> >You link one DAI in I2S mode to the bcm2835 and the other DAI
> >in DSP (or whatever mode you are using) to the cs42xx8.
> >If you model it this way you no longer work against ALSA and
> >you can stop adding hacks to existing drivers.
> From the beginning, I completely agree with you when you take the two
> problems apart:
> - for the timing problem : model properly the converter as a codec
> - for the encapsulation problem : do the encapsulation / packing with a
> userspace plugin
> But when you take the whole together, the plugin part seems completely
> As the whole is properly modeled, if we could have a simple solution to
> relax the channel numbers constraint on the I2S on the higher part of the
> stack all will "magically" work with little effort/complexity.
> Otherwise, the user-space packer would have to do a lot of more than packing
> : interact with private machine driver controls to manage all the channels
> and the machine driver will need to forward /translate some parts to
> directly drive the final (cs42xx8) codec. An potentially if such hardware
> continue to pop-up, each machine driver will need it's own user-space
Quite on the contrary. You'd need to add the channel relaxing constraint
to all existing I2S drivers to get this working. And I don't see a need
to artificially restrict that to a specific codec (cs42xx8) and a specific
number of channels. Why only 8 channels, why not 4 or 384?
Doing it as an ALSA plugin makes it reusable with existing drivers and
The idea behind these modular components (plugins, codecs, dais) is to
create reusable components and you don't have to add identical code
to all other drivers when you add some new functionality.
Actually, if you add a new feature, you need to have very good reasons
to restrict it to a specific driver or change all existing drivers.
If you can implement that feature in a generic way without touching
existing code it's often the better solution.
Matt is trying to tunnel multichannel PCM over a 2-channel PCM link running
at a higher samplerate. I've described a way how this is possible without
modifying current code. There are certainly other, probably better,
ways to do that. This was a first quick idea how it could be done and I
still think it's not too bad.
All the plugin has to do is expose a multi-channel PCM and configure the
hw/backend PCM to 2 channels at a higher samplerate (all of which
current drivers are already capable of). The plugin settings determine
the number of channels, channel map, samplerate factor etc. That same
plugin can also be used with other "unpacking codecs" and other channel
numbers - you just need to change the plugin configuration in your alsa
card conf and tell it you have 4 (or whatever) channels.
If you need interaction with the backend codec (Matt's "IC"/FPGA) that
does PCM unpacking to multichannel you can do that for example in the plugin
or in the alsa card.conf via the hooks plugin and alsa controls.
> Packing DSD in PCM (DoP) require zero kernel knowledge and could be fully
> implemented in user-space as we don't change the number of channel
> assumption of the data part of the format. But here, we simply want the DSP
> A data semantic with the I2S hardware bus timing.
Yes, and this is what Matt's codec (FPGA) is doing. It's creating that
semantic, together with the machine driver and the plugin to set it all up.
> I'm a complete newbie to ASoC but I take part to this tread to learn as I
> hate to see how badly all diy and amateur audio hw are integrated with
> Alsa/ASoC/Linux and so never go upstream. On a professional/commercial dev,
> you would never take this ... convoluted I2S multi channel path.
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