[alsa-devel] [PATCH 0/3] ASoC: Enable a new IC master mode: bcm2835<=>IC<=>cs42xx8

Matthias Reichl hias at horus.com
Sun Feb 26 14:16:48 PST 2017

On Mon, Feb 27, 2017 at 07:21:13AM +1100, Matt Flax wrote:
> On 27/02/17 01:49, Matthias Reichl wrote:
> >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
> >>the
> >>bcm2835 was run in DSP mode with a codec master (rather then IC master) and
> >>that
> >>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.
> All configuration is done in a machine driver, which I haven't submitted.
> The machine driver's dai_fmt is as follows :
> >>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
> >>master
> >>(which we know produces bit offset and random channel swapping when a codec
> >>is
> >>master).
> >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
> >somewhere.
> Clocks are set up in the machine driver as well.
> >
> >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.
> >
> >
> Thanks, this is actually very constructive advice. Let me try to understand
> what
> you are suggesting here... I am confused about how to set bit depth
> correctly for
> the codec when I play to the IC chip in this setup... lets walk me through
> this one...
> I implement a new codec which matches this IC. The IC can be setup to be
> master in
> its dai fmt. The sound card shows up with two devices, only the IC device
> can be used to play
> and record.

Actually you don't get 2 PCMs from that in userspace, you are only
telling ALSA that your stream is routed through another codec inbetween.

> Say I use aplay to play to the first device (the IC chip) it does hw_params
> and clocks are set.
> But the second device (the codec) never gets hw_params executed ? If I
> select 16 or 24 bits,
> it never knows ... is that right ? If so, how do we solve this problem ?

Have a look at the end of Documentation/sound/alsa/soc/DPCM.txt
where codec<->codec links are explained.

You can setup the stream parameters for the "IC"<->"real codec" link
with snd_soc_dai_link.params.

You can do that for example in your machine driver in hwparams:
when hw_params is called eg with 2 channels 192kHz just set the
dai link parameters of the codec to 8 channels 48kHz.

Maybe using dynamic pcm (which offers a be_hw_params_fixup hook)
is the more appropriate way to do that, but I can't tell for sure
as I never used it and am not familiar with it.

so long,


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