[alsa-devel] Question about your DSP topic branch
plai at codeaurora.org
Tue Mar 15 08:08:39 CET 2011
In the DSP routing scheme, what is expectation of back-end CPU/Codec
drivers in term of determining what hardware parameters especially
channel mode to use? DSP in question is capable of resampling
, down mixing so hardware configuration of front-end DAI does not need
to match one of back-end.
Here are the scenarios that I need clarification
1. If there are two front-ends routed to same backend, which front-end
hardware parameters should backend DAI be based on? For example, one
front-end is MONO and another front is stereo.
2. Depending on device mode/use case, I would like to configure BE to
different channel mode irrespective of front-end configuration(i.e
configuring back-end to handset mode). Where is the hook to do so under
ASoC DSP framework?
On 1/27/2011 3:41 PM, Patrick Lai wrote:
> On 1/25/2011 8:46 AM, Liam Girdwood wrote:
>> Hi Patrick,
>> CCing in Mark and alsa-devel at alsa-project.org (preferred list)
>> On Mon, 2011-01-24 at 23:01 -0800, Patrick Lai wrote:
>>> Hi Liam,
>>> I have two more questions about your DSP topic branch
>>> 7. I see in sdp4430.c, SDP4430 MODEM front-end dailink no_host_mode is
>>> set to SND_SOC_DAI_LINK_NO_HOST. What is the purpose of no_host_mode?
>> No host mode means no audio is transferred to the host CPU in this case.
>> i.e. the MODEM routes audio directly between the DSP and mic/speaker.
>> This flags also tells the ASoC core that no DMA will be required, hence
>> the code in the DMA branch will not start any DMA. This part had to be
>> cleaned up for upstream.
>>> Is it for use case that two physical devices can exchange audio data
>>> without host-processor intervention? If so, if user-space application
>>> tries to write to PCM buffer, will framework reject the buffer?
>> Yes, that's correct. The PCM core will also not complain here when it
>> receive no data either.
> I experimented NO_HOST option and found platform driver pcm copy
> function is called. Is it part of the clean up you were talking about?
>>> 8. I see there is dmic codec(dmic.c) under sound/soc/codec which is
>>> pretty much just a dummy codec driver. I supposed the configuration of
>>> DMIC is done in other driver. Would it be better if we could have
>>> something like fixed voltage regulator? So, there is no need to
>>> duplicate the effort.
>> Yeah, this is a generic DMIC driver. It's designed to have a very wide
>> coverage and should cover most DMICs out there. So it should also be
>> able to fit into your architecture too.
>>> Look forward to seeing your reply soon
>>> On 1/6/2011 3:39 PM, Patrick Lai wrote:
>>>> Hi Liam,
>>>> I sync to your kernel DSP topic branch two days back in attempt to
>>>> understand the up-coming ASOC DSP design. I have few questions to get
>>>> clarification from you.
>>>> 1. In the sdp4430.c, there are both FE dai link and BE dai link have
>>>> omap-aess-audio as platform driver. How is omap-aess-audio platform
>>>> driver used in both front-end and backend?
>> The MODEM and Low Power (LP) Front Ends (FE) use the AESS platform
>> driver since they do not require DMA, whilts the other FE's use the
>> normal DMA platform driver since they do require DMA.
> PCM functions inside omap-abe-dsp.c seem to do no-op if dai ID is not
> MODEM or LP. I guess reusing omap-aess-audio platform driver for the
> back-end DAI link serves purpose of architecture requirement that each
> DAI-LINK must have platform driver. Do I get the right impression?
>>>> 2. Front-end dai with stream name "Multimedia" has platform driver as
>>>> omap-pcm-audio which is the DMA based. This front-end dai is mapped to
>>>> backend (i.e PDM-DL1) with platform driver as omap-aess-audio.
>>>> Omap-aess-audio looks to me is DSP platform driver. If a stream is DMA
>>>> based, it seems strange to have DSP-based backend.
>> The DMA is used to send the PCM data from the ALSA PCM device to the DSP
>>>> 3. To best of my knowledge, I see in omap-abe.c which is the front end
>>>> implementation. Playback_trigger() seems to go ahead enable all
>>>> back-ends linked to the given front-end. However, front end may not be
>>>> routed to all back-ends. Where in the code to make sure BE is only
>>>> activated when a stream is being routed to that particular back-end?
>> This is all done in soc-dsp.c now. We use the DAPM graph to work out all
>> valid routes from FE to BE and vice versa.
> I experimented dynamic routing. If issuing mixer command to route AIF_IN
> to AIF_OUT through mixer widget before starting playback, it works.
> Otherwise, I get I/O error from aplay. I think it's acceptable to
> require application to setup path before start of playback. For device
> switch case, is it mandatory to route stream to the mixer of BE
> DAI-LINK2 before derouting stream out of mixer of BE DAI-LINK1?
>>>> 4. omap-abe.c manages activation of BE DAIS and omap-abe-dsp.c manages
>>>> routing of front-end to back-end DAPM widgets and routing map. Am I
>> Yes, although the routing management is now all in soc-dsp.c
>>> This leads to previous question. How are two drivers working
>>>> together to make sure BEs are only activated if Front-end has been
>>>> routed to them?
>> soc-dsp.c now marshals all the PCM ops and makes sure that only valid
>> paths have their DAIs activated.
>>>> 5. Is there mechanism for front-end to switch between DMA and DSP
>>>> platform driver? It looks to me that mapping of frond-end and platform
>>>> driver is predetermined based on use case. For example, HDMI goes
>>>> through DMA as legacy dailink.
>> There is no way to dynamically switch platform drivers atm, but this can
>> be solved by having a mutually exclusive FE for each use case.
>>>> 6. struct abe_data has array member called dapm. It looks to me that
>>>> this array simply tracks dapm components status but I don't see it's
>>>> really being used in meaningful way in omap-abe-adsp.c.
>> It used by the OMAP4 ABE to work out the OPP power level and to work out
>> the routing between FE and BE.
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