Re: [alsa-devel] [PATCH v2 3/5] ASoC: core: add support to snd_soc_dai_get_sdw_stream() - Pierre-Louis Bossart

From: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
To: Charles Keepax <ckeepax@opensource.cirrus.com>,
	Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: devicetree@vger.kernel.org, alsa-devel@alsa-project.org,
	bgoswami@codeaurora.org, plai@codeaurora.org,
	linux-kernel@vger.kernel.org, lgirdwood@gmail.com,
	Vinod Koul <vkoul@kernel.org>,
	robh+dt@kernel.org, Mark Brown <broonie@kernel.org>,
	spapothi@codeaurora.org
Subject: Re: [alsa-devel] [PATCH v2 3/5] ASoC: core: add support to snd_soc_dai_get_sdw_stream()
Date: Thu, 10 Oct 2019 09:01:42 -0500	[thread overview]
Message-ID: <22eff3aa-dfd6-1ee5-8f22-2af492286053@linux.intel.com> (raw)
In-Reply-To: <20191010120337.GB31391@ediswmail.ad.cirrus.com>

>>> It's been a while since this thread started, and I still don't
>>> quite get the concepts or logic.
>>>
>>> First, I don't understand what the problem with "aux" devices is.
>>> All the SoundWire stuff is based on the concept of DAI, so I guess
>>> I am
>>
>> That is the actual problem! Some aux devices does not have dais.
>>
> 
> Usually aux devices are used for things like analog amplifiers that
> clearly don't have a digital interface, thus it really makes no sense
> to have a DAI link connecting them. So I guess my question here
> would be what is the thinking on making the "smart amplifier" dailess?
> It feels like having a CODEC to CODEC DAI between the CODEC and
> the AMP would be a more obvious way to connect the two devices
> and would presumably avoid the issues being discussed around the
> patch.

Ah, now I get it, I missed the point about not having DAIs for the 
amplifier.

I will second Charles' point, the code you have in the machine driver at 
[1] gets a SoundWire stream context from the SLIMbus codec dai. It's a 
bit odd to mix layers this way.

And if I look at the code below, taken from [1], if you have more than 
one codec, then your code looks problematic: data->sruntime would be 
overriden and you'd use the info from the last codec dai on the dailink...

case SLIMBUS_0_RX...SLIMBUS_6_TX:
   for (i = 0 ; i < dai_link->num_codecs; i++) {
     [snip]
     data->sruntime[cpu_dai->id] =
	snd_soc_dai_get_sdw_stream(rtd->codec_dais[i], 0); << same destination 
for multiple codec_dais...

If you keep the amp dai-less, then the stream concept should be somehow 
allocated on the master (or machine) side and passed to the codec dais 
that are associated to the same 'stream'.

[1] 
https://git.linaro.org/people/srinivas.kandagatla/linux.git/tree/sound/soc/qcom/db845c.c?h=release/db845c/qcomlt-5.2 

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