From: Patrick Lai <plai@codeaurora.org>
To: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: "alsa-devel@alsa-project.org" <alsa-devel@alsa-project.org>,
Mark <broonie@opensource.wolfsonmicro.com>
Subject: Question about your DSP topic branch - hw param fix up query
Date: Thu, 31 Mar 2011 10:45:23 -0700 [thread overview]
Message-ID: <4D94BDB3.4050903@codeaurora.org> (raw)
In-Reply-To: <1301512193.3375.31.camel@odin>
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?
Thanks
Patrick
>
> _______________________________________________
> Alsa-devel mailing list
> Alsa-devel@alsa-project.org
> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
--
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next prev parent reply other threads:[~2011-03-31 17:45 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 ` Patrick Lai [this message]
2011-03-31 21:35 ` Question about your DSP topic branch - hw param fix up query Liam Girdwood
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