[alsa-devel] [PATCH] ASoC: bcm2835: Add 8 channel (multitrack) capability

[Matt Flax]

On 09/02/17 08:13, Matt Flax wrote:
>
>
> On 09/02/17 05:54, Matthias Reichl wrote:
>> On Wed, Feb 08, 2017 at 06:28:35PM +0000, Mark Brown wrote:
>>> On Tue, Feb 07, 2017 at 10:09:36AM +1100, Matt Flax wrote:
>>>
>>>>       case SND_SOC_DAIFMT_CBS_CFM:
>>>>           clk_set_rate(dev->clk, sampling_rate * bclk_ratio);
>>>> +    case SND_SOC_DAIFMT_CBM_CFS:
>>> Is this fall through deliberate?
>>>
>>>> +    /* Default data delay to 1 bit.
>>>> +       In I2S mode, we must have 2 channels */
>>>>       switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
>>>>       case SND_SOC_DAIFMT_I2S:
>>>> +        if (params_channels(params) != 2)
>>>> +            return -EINVAL;
>>>> +    case SND_SOC_DAIFMT_DSP_A:
>>>> +    case SND_SOC_DAIFMT_DSP_B:
>>>>           data_delay = 1;
>>>>           break;
>>>>       default:
>> Matt, could you please include linux-rpi-kernel at lists.infradead.org
>> in your emails?
> I have joined that list now. It was included originally, but wasn't 
> accepting my posts.
>> I fail to see the part where DSP modes are actually set up in
>> the hardware. bcm2835 still seems to be operating in 2-channel
>> stereo I2S mode, i.e. no real frame sync information at the
>> hardware level.
> From the SoC's perspective I agree with you. There is frame 
> synchronisation at the hardware level, implemented in an master FPGA. 
> This starts to hit at a lack of functionality in ALSA ... I will 
> discuss more below.
>> If all you do is adding code to pretend the bcm2835 could do
>> multichannel modes wouldn't it be easier to implement that as
>> a userspace alsa plugin?
>>
>>
> I am not familiar with how to implement all of this with a plugin ? 
> Could you give me a little hand in describing that further ? That 
> would mean that an asoundrc needs to be used to defined to make the 
> system usable ? Is it something which does the unpacking for us in 
> user space ? If this happens in user space is there extra cost/latency ?
>

You know, I am genuinely interested in your concept and still invite an 
example of your creativity, however ...
The more I think about this approach, the concept of pushing the support 
of hardware into user space the more I disagree with it. My 
understanding is that the Linux Kernel is there to support hardware. The 
concept of pushing hardware support into user space doesn't seem right.

As I have pointed out below, there are missing things in ALSA and as 
Mark previously pointed out "this is a thing". What I understand is that 
this hardware is a thing and has been thought of before - this happens 
to be a hardware implementation of this "thing" which ALSA doesn't 
currently have the capacity to support (e.g. an ASIC/FPGA which is 
mater, not the SoC nor the Codec).

I remember back in the '90s when ALSA was started - I witnessed its 
birth. ALSA was started because of inadequacies of OSS. I truly don't 
believe that we need a new sound system for Linux as of yet. I also 
don't believe that because ALSA has these inadequacies (which I mention 
below) that we need to start afresh. I would personally put effort into 
this part of ALSA if I had the money to support myself whilst I did it - 
but I don't. So for now, I am trying to make do with ALSA as best I can. 
I am trying to put the necessary support for existing hardware into ALSA 
in its current state and form - in the best possible manner. So please 
lets continue with support for this hardware in the kernel.

> I would like to bring up another topic here.
>
> In my opinion some of these changes we are making in this general 
> thread are only really window dressing.
>
> We have 4 ways of setting up master, however all of them assume that 
> either the codec or the SoC is master. None of them allow for 
> intermediate digital logic between the two.
>
> In this case there is a FPGA which is matching the system differences 
> between the Codec and the SoC. In actual fact, the FPGA needs to be 
> master - a fifth mode.
>
> A similar problem exists when you are using a sample rate converter 
> chip. For example, the DAC and ADC are running at different sample 
> rates. In this case ALSA can't represent both of the sample rates. For 
> that reason, the ADCs and the DACs have to be hard coded - it is nasty.
>
> The only solution for me is to use snd_soc_dai_set_fmt in the machine 
> driver to instruct both to enter slave mode. For what it is worth, I 
> can also
>
> In my opinion there is nothing wrong with making hardware level 
> introductions, such as an ASIC/FPGA to implement the hardware. I 
> accept the inflexibility of ALSA w.r.t. this type of situation, 
> however the real fix is to adjust the core of ALSA. Hardware ASICS and 
> FPGAs which are intermediatries between codecs and SoCs exist and are 
> used in industry.
>
> This happens to be one of those cases.
>
> thanks
> Matt
>
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