From: Liam Girdwood <lrg@slimlogic.co.uk>
To: Patrick Lai <plai@codeaurora.org>
Cc: "alsa-devel@alsa-project.org" <alsa-devel@alsa-project.org>,
Mark <broonie@opensource.wolfsonmicro.com>
Subject: Re: Question about your DSP topic branch - hw param fix up query
Date: Thu, 31 Mar 2011 22:35:31 +0100 [thread overview]
Message-ID: <1301607331.3549.215.camel@odin> (raw)
In-Reply-To: <4D94BDB3.4050903@codeaurora.org>
On Thu, 2011-03-31 at 10:45 -0700, Patrick Lai wrote:
> Change subject
>
> > static int dmic_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
> > struct snd_pcm_hw_params *params)
> > {
> > struct snd_interval *rate = hw_param_interval(params,
> > SNDRV_PCM_HW_PARAM_RATE);
> >
> > /* The ABE will covert the FE rate to 96k */
> > rate->min = rate->max = 96000;
> >
> > snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT -
> > SNDRV_PCM_HW_PARAM_FIRST_MASK],
> > SNDRV_PCM_FORMAT_S32_LE);
> > return 0;
> > }
> Liam,
>
> For sample rate conversion case, supplying back-end hw_params_fixup
> function would work. However, I am looking for run-time configuration of
> back-end channel mode. For scenario of multi-channel microphone input, I
> need to have a mean to specify channel mode of back-end depending on
> algorithm running on DSP while front-end channel mode is mono. Is there
> a hook in the framework to do that?
>
Not atm, but it does sound like we need to add such a hook.
I'll add this to the conference agenda.
Thanks
Liam
prev parent reply other threads:[~2011-03-31 21:35 UTC|newest]
Thread overview: 6+ messages / expand[flat|nested] mbox.gz Atom feed top
2011-03-30 11:17 ALSA hw sample rate conversion support Aisheng Dong
2011-03-30 13:27 ` Clemens Ladisch
2011-03-30 15:32 ` Koul, Vinod
2011-03-30 19:09 ` Liam Girdwood
2011-03-31 17:45 ` Question about your DSP topic branch - hw param fix up query Patrick Lai
2011-03-31 21:35 ` Liam Girdwood [this message]
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