From mboxrd@z Thu Jan 1 00:00:00 1970 From: Liam Girdwood Subject: Re: [PATCH] ASoC: fsi: Add specified ID for soc-audio Date: Sat, 17 Jul 2010 18:42:37 +0100 Message-ID: <1279388557.3070.3.camel@odin> References: Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from mail-wy0-f179.google.com (mail-wy0-f179.google.com [74.125.82.179]) by alsa0.perex.cz (Postfix) with ESMTP id 83470246E1 for ; Sat, 17 Jul 2010 19:42:43 +0200 (CEST) Received: by wyf19 with SMTP id 19so3041997wyf.38 for ; Sat, 17 Jul 2010 10:42:42 -0700 (PDT) In-Reply-To: List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Sender: alsa-devel-bounces@alsa-project.org Errors-To: alsa-devel-bounces@alsa-project.org To: Kuninori Morimoto Cc: Linux-ALSA , Mark Brown List-Id: alsa-devel@alsa-project.org On Fri, 2010-07-16 at 19:51 +0900, Kuninori Morimoto wrote: > Specified ID is necessary, when some codecs are used with FSI. > > Signed-off-by: Kuninori Morimoto > --- > Mark. Thank you about your advice. > it solved my issue. > This patch is v2 of "ASoC: FSI - codecs settings" > > include/sound/sh_fsi.h | 3 +++ > sound/soc/sh/fsi-ak4642.c | 4 ++-- > sound/soc/sh/fsi-da7210.c | 4 ++-- > 3 files changed, 7 insertions(+), 4 deletions(-) > > diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h > index c022736..6463fd6 100644 > --- a/include/sound/sh_fsi.h > +++ b/include/sound/sh_fsi.h > @@ -12,6 +12,9 @@ > * published by the Free Software Foundation. > */ > > +#define FSI_PORT_A 0 > +#define FSI_PORT_B 1 > + > /* flags format > > * 0xABCDEEFF > diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c > index be01854..2c61ed2 100644 > --- a/sound/soc/sh/fsi-ak4642.c > +++ b/sound/soc/sh/fsi-ak4642.c > @@ -38,7 +38,7 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) > static struct snd_soc_dai_link fsi_dai_link = { > .name = "AK4642", > .stream_name = "AK4642", > - .cpu_dai = &fsi_soc_dai[0], /* fsi */ > + .cpu_dai = &fsi_soc_dai[FSI_PORT_A], > .codec_dai = &ak4642_dai, > .init = fsi_ak4642_dai_init, > .ops = NULL, > @@ -62,7 +62,7 @@ static int __init fsi_ak4642_init(void) > { > int ret = -ENOMEM; > > - fsi_snd_device = platform_device_alloc("soc-audio", -1); > + fsi_snd_device = platform_device_alloc("soc-audio", FSI_PORT_A); > if (!fsi_snd_device) > goto out; > > diff --git a/sound/soc/sh/fsi-da7210.c b/sound/soc/sh/fsi-da7210.c > index 33b4d17..5774449 100644 > --- a/sound/soc/sh/fsi-da7210.c > +++ b/sound/soc/sh/fsi-da7210.c > @@ -33,7 +33,7 @@ static int fsi_da7210_init(struct snd_soc_codec *codec) > static struct snd_soc_dai_link fsi_da7210_dai = { > .name = "DA7210", > .stream_name = "DA7210", > - .cpu_dai = &fsi_soc_dai[1], /* FSI B */ > + .cpu_dai = &fsi_soc_dai[FSI_PORT_B], > .codec_dai = &da7210_dai, > .init = fsi_da7210_init, > }; > @@ -56,7 +56,7 @@ static int __init fsi_da7210_sound_init(void) > { > int ret; > > - fsi_da7210_snd_device = platform_device_alloc("soc-audio", -1); > + fsi_da7210_snd_device = platform_device_alloc("soc-audio", FSI_PORT_B); > if (!fsi_da7210_snd_device) > return -ENOMEM; > Acked-by: Liam Girdwood -- Freelance Developer, SlimLogic Ltd ASoC and Voltage Regulator Maintainer. http://www.slimlogic.co.uk