[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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