From mboxrd@z Thu Jan 1 00:00:00 1970 From: Nicolin Chen Subject: Re: [PATCH 2/2] ASoC: fsl_spdif: Add support for output sample rates 96kHz and 192kHz Date: Mon, 16 Jun 2014 10:33:59 +0800 Message-ID: <20140616023359.GC21550@MrMyself> References: <1402876602-2942-1-git-send-email-anssi.hannula@iki.fi> <1402876602-2942-2-git-send-email-anssi.hannula@iki.fi> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Return-path: Received: from na01-bl2-obe.outbound.protection.outlook.com (mail-bl2lp0207.outbound.protection.outlook.com [207.46.163.207]) by alsa0.perex.cz (Postfix) with ESMTP id F11AF2652E7 for ; Mon, 16 Jun 2014 04:51:43 +0200 (CEST) Content-Disposition: inline In-Reply-To: <1402876602-2942-2-git-send-email-anssi.hannula@iki.fi> List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: alsa-devel-bounces@alsa-project.org To: Anssi Hannula Cc: alsa-devel@alsa-project.org, Mark Brown List-Id: alsa-devel@alsa-project.org On Mon, Jun 16, 2014 at 02:56:42AM +0300, Anssi Hannula wrote: > Add support for the output sample rates 96kHz and 192kHz. > > Tested with a Cubox-i imx6 system and an Onkyo TX-SR607 receiver. > > Signed-off-by: Anssi Hannula > Cc: Nicolin Chen Great, thank you. Acked-by: Nicolin Chen > --- > > Unless there was some special reason to not have this? > > It seems to work at least. > > sound/soc/fsl/fsl_spdif.c | 14 +++++++++++--- > sound/soc/fsl/fsl_spdif.h | 8 ++++++-- > 2 files changed, 17 insertions(+), 5 deletions(-) > > diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c > index b912d45a2a4c..71f6a117f46e 100644 > --- a/sound/soc/fsl/fsl_spdif.c > +++ b/sound/soc/fsl/fsl_spdif.c > @@ -96,7 +96,7 @@ struct fsl_spdif_priv { > struct platform_device *pdev; > struct regmap *regmap; > bool dpll_locked; > - u16 txrate[SPDIF_TXRATE_MAX]; > + u32 txrate[SPDIF_TXRATE_MAX]; > u8 txclk_df[SPDIF_TXRATE_MAX]; > u8 sysclk_df[SPDIF_TXRATE_MAX]; > u8 txclk_src[SPDIF_TXRATE_MAX]; > @@ -390,6 +390,14 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, > rate = SPDIF_TXRATE_48000; > csfs = IEC958_AES3_CON_FS_48000; > break; > + case 96000: > + rate = SPDIF_TXRATE_96000; > + csfs = IEC958_AES3_CON_FS_96000; > + break; > + case 192000: > + rate = SPDIF_TXRATE_192000; > + csfs = IEC958_AES3_CON_FS_192000; > + break; > default: > dev_err(&pdev->dev, "unsupported sample rate %d\n", sample_rate); > return -EINVAL; > @@ -1046,7 +1054,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, > struct clk *clk, u64 savesub, > enum spdif_txrate index, bool round) > { > - const u32 rate[] = { 32000, 44100, 48000 }; > + const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; > bool is_sysclk = clk == spdif_priv->sysclk; > u64 rate_ideal, rate_actual, sub; > u32 sysclk_dfmin, sysclk_dfmax; > @@ -1105,7 +1113,7 @@ out: > static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, > enum spdif_txrate index) > { > - const u32 rate[] = { 32000, 44100, 48000 }; > + const u32 rate[] = { 32000, 44100, 48000, 96000, 192000 }; > struct platform_device *pdev = spdif_priv->pdev; > struct device *dev = &pdev->dev; > u64 savesub = 100000, ret; > diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h > index 16fde4b927d3..5f5a3c41db25 100644 > --- a/sound/soc/fsl/fsl_spdif.h > +++ b/sound/soc/fsl/fsl_spdif.h > @@ -164,8 +164,10 @@ enum spdif_txrate { > SPDIF_TXRATE_32000 = 0, > SPDIF_TXRATE_44100, > SPDIF_TXRATE_48000, > + SPDIF_TXRATE_96000, > + SPDIF_TXRATE_192000, > }; > -#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_48000 + 1) > +#define SPDIF_TXRATE_MAX (SPDIF_TXRATE_192000 + 1) > > > #define SPDIF_CSTATUS_BYTE 6 > @@ -175,7 +177,9 @@ enum spdif_txrate { > > #define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \ > SNDRV_PCM_RATE_44100 | \ > - SNDRV_PCM_RATE_48000) > + SNDRV_PCM_RATE_48000 | \ > + SNDRV_PCM_RATE_96000 | \ > + SNDRV_PCM_RATE_192000) > > #define FSL_SPDIF_RATES_CAPTURE (SNDRV_PCM_RATE_16000 | \ > SNDRV_PCM_RATE_32000 | \ > -- > 1.8.4.5 >