[alsa-devel] [PATCH 3/3] ASoC: cs42l73: Use DAPM routes to hook AIF widgets to DAI's

Brian Austin brian.austin at cirrus.com
Wed May 9 19:18:13 CEST 2012


On Wed, 9 May 2012, Brian Austin wrote:

> Signed-off-by: Brian Austin <brian.austin at cirrus.com>
> ---
> sound/soc/codecs/cs42l73.c |   15 +++++++++++++++
> 1 file changed, 15 insertions(+)
>
> diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c
> index 1cf2206..d8b325c 100644
> --- a/sound/soc/codecs/cs42l73.c
> +++ b/sound/soc/codecs/cs42l73.c
> @@ -773,6 +773,14 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
> 	{"HL Left Mixer", NULL, "VSPIN"},
> 	{"HL Right Mixer", NULL, "VSPIN"},
>
> +	{"ASPINL", NULL, "ASP Playback"},
> +	{"ASPINM", NULL, "ASP Playback"},
> +	{"ASPINR", NULL, "ASP Playback"},
> +	{"XSPINL", NULL, "XSP Playback"},
> +	{"XSPINM", NULL, "XSP Playback"},
> +	{"XSPINR", NULL, "XSP Playback"},
> +	{"VSPIN", NULL, "VSP Playback"},
> +
> 	/* Capture Paths */
> 	{"MIC1", NULL, "MIC1 Bias"},
> 	{"PGA Left Mux", "Mic 1", "MIC1"},
> @@ -819,6 +827,13 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = {
>
> 	{"VSPOUTL", NULL, "VSPL Output Mixer"},
> 	{"VSPOUTR", NULL, "VSPR Output Mixer"},
> +
> +	{"ASP Capture", NULL, "ASPOUTL"},
> +	{"ASP Capture", NULL, "ASPOUTR"},
> +	{"XSP Capture", NULL, "XSPOUTL"},
> +	{"XSP Capture", NULL, "XSPOUTR"},
> +	{"VSP Capture", NULL, "VSPOUTL"},
> +	{"VSP Capture", NULL, "VSPOUTR"},
> };
I forgot to add the other part in the SND_SOC_DAPM_AIF_IN and 
SND_SOC_DAPM_AIF_OUT Widgets.

I'll resend this one, sorry



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