From mboxrd@z Thu Jan 1 00:00:00 1970 From: Kuninori Morimoto Subject: [PATCH 2/3] ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops Date: Mon, 15 Mar 2010 18:10:50 +0900 Message-ID: References: Mime-Version: 1.0 (generated by SEMI 1.14.6 - "Maruoka") Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from mail07.idc.renesas.com (mail.renesas.com [202.234.163.13]) by alsa0.perex.cz (Postfix) with ESMTP id 7889F103869 for ; Tue, 23 Mar 2010 08:27:36 +0100 (CET) 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: Mark Brown Cc: Linux-ALSA List-Id: alsa-devel@alsa-project.org Signed-off-by: Kuninori Morimoto --- sound/soc/codecs/ak4642.c | 69 +++++++++++++++++++++++++++++--------------- sound/soc/sh/fsi-ak4642.c | 4 ++ 2 files changed, 49 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index d5bd4ca..3452bd7 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -80,6 +80,17 @@ #define AK4642_CACHEREGNUM 0x25 +/* PW_MGMT2 */ +#define HPMTN (1 << 6) +#define PMHPL (1 << 5) +#define PMHPR (1 << 4) +#define MS (1 << 3) /* master/slave select */ +#define MCKO (1 << 1) +#define PMPLL (1 << 0) + +#define PMHP_MASK (PMHPL | PMHPR) +#define PMHP PMHP_MASK + /* MD_CTL1 */ #define PLL3 (1 << 7) #define PLL2 (1 << 6) @@ -87,6 +98,9 @@ #define PLL0 (1 << 4) #define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0) +#define BCKO_MASK (1 << 3) +#define BCKO_64 BCKO_MASK + struct snd_soc_codec_device soc_codec_dev_ak4642; /* codec private data */ @@ -188,9 +202,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. - * - * Example code use 0x39, 0x79 value for 0x01 address, - * But we need MCKO (0x02) bit now */ ak4642_write(codec, 0x05, 0x27); ak4642_write(codec, 0x0f, 0x09); @@ -200,8 +211,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, ak4642_write(codec, 0x0a, 0x28); ak4642_write(codec, 0x0d, 0x28); ak4642_write(codec, 0x00, 0x64); - ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */ - ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */ + snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); + snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { /* * start stereo input @@ -238,8 +249,8 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, if (is_play) { /* stop headphone output */ - ak4642_write(codec, 0x01, 0x3b); - ak4642_write(codec, 0x01, 0x0b); + snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0); + snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0); ak4642_write(codec, 0x00, 0x40); ak4642_write(codec, 0x0e, 0x11); ak4642_write(codec, 0x0f, 0x08); @@ -284,10 +295,37 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 data; + u8 bcko; + + data = MCKO | PMPLL; /* use MCKO */ + bcko = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + data |= MS; + bcko = BCKO_64; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + snd_soc_update_bits(codec, PW_MGMT2, MS, data); + snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); + + return 0; +} + static struct snd_soc_dai_ops ak4642_dai_ops = { .startup = ak4642_dai_startup, .shutdown = ak4642_dai_shutdown, .set_sysclk = ak4642_dai_set_sysclk, + .set_fmt = ak4642_dai_set_fmt, }; struct snd_soc_dai ak4642_dai = { @@ -366,23 +404,6 @@ static int ak4642_init(struct ak4642_priv *ak4642) goto reg_cache_err; } - /* - * clock setting - * - * Audio I/F Format: MSB justified (ADC & DAC) - * BICK frequency at Master Mode: 64fs - * MCKO: Enable - * Sampling Frequency: 44.1kHz - * - * This operation came from example code of - * "ASAHI KASEI AK4642" (japanese) manual p89. - * - * please fix-me - */ - ak4642_write(codec, 0x01, 0x08); - ak4642_write(codec, 0x05, 0x27); - ak4642_write(codec, 0x04, 0x0a); - return ret; reg_cache_err: diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index c0207dc..be01854 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -26,6 +26,10 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) { int ret; + ret = snd_soc_dai_set_fmt(&ak4642_dai, SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_sysclk(&ak4642_dai, 0, 11289600, 0); return ret; -- 1.6.3.3