From patchwork Wed May 12 00:45:33 2021 Content-Type: text/plain; charset="utf-8" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit X-Patchwork-Submitter: Kuninori Morimoto X-Patchwork-Id: 435690 Return-Path: X-Spam-Checker-Version: SpamAssassin 3.4.0 (2014-02-07) on aws-us-west-2-korg-lkml-1.web.codeaurora.org X-Spam-Level: X-Spam-Status: No, score=-13.8 required=3.0 tests=BAYES_00,DKIM_SIGNED, DKIM_VALID, HEADER_FROM_DIFFERENT_DOMAINS, INCLUDES_CR_TRAILER, INCLUDES_PATCH, MAILING_LIST_MULTI, SPF_HELO_NONE, SPF_PASS autolearn=ham autolearn_force=no version=3.4.0 Received: from mail.kernel.org (mail.kernel.org [198.145.29.99]) by smtp.lore.kernel.org (Postfix) with ESMTP id 623CAC433B4 for ; Wed, 12 May 2021 00:47:05 +0000 (UTC) Received: from alsa0.perex.cz (alsa0.perex.cz [77.48.224.243]) (using TLSv1.2 with cipher ECDHE-RSA-AES256-GCM-SHA384 (256/256 bits)) (No client certificate requested) by mail.kernel.org (Postfix) with ESMTPS id 7632E6162B for ; Wed, 12 May 2021 00:47:04 +0000 (UTC) DMARC-Filter: OpenDMARC Filter v1.3.2 mail.kernel.org 7632E6162B Authentication-Results: mail.kernel.org; dmarc=none (p=none dis=none) header.from=renesas.com Authentication-Results: mail.kernel.org; spf=pass smtp.mailfrom=alsa-devel-bounces@alsa-project.org Received: from alsa1.perex.cz (alsa1.perex.cz [207.180.221.201]) (using TLSv1.2 with cipher AECDH-AES256-SHA (256/256 bits)) (No client certificate requested) by alsa0.perex.cz (Postfix) with ESMTPS id E05F618BE; Wed, 12 May 2021 02:46:12 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa0.perex.cz E05F618BE DKIM-Signature: v=1; a=rsa-sha256; c=relaxed/simple; d=alsa-project.org; s=default; t=1620780423; bh=Ld+pUBG9g8XVJGBlyTkSfdPXONb5I97rrVAswYdoSj0=; h=Date:From:Subject:To:In-Reply-To:References:Cc:List-Id: List-Unsubscribe:List-Archive:List-Post:List-Help:List-Subscribe: From; b=QRNAW7uDbYwNHknPoKhAkPA4HtNBfokEFP/QHpaTfiac8hW3VAWR6r/cUBqSoj1D0 zAd4+t8IOK/gOClb+NtfVg/pYf5zb4V6TYtBIMd/aLWMSxOMxyiwyOJmCXhz6M6NRW hOEXvc2Xk9DHH9JS9cCFpwi7aitwr3u4jP17yuVU= Received: from alsa1.perex.cz (localhost.localdomain [127.0.0.1]) by alsa1.perex.cz (Postfix) with ESMTP id 0A2ACF80424; Wed, 12 May 2021 02:45:43 +0200 (CEST) Received: by alsa1.perex.cz (Postfix, from userid 50401) id 5927EF80424; Wed, 12 May 2021 02:45:41 +0200 (CEST) Received: from relmlie6.idc.renesas.com (relmlor2.renesas.com [210.160.252.172]) by alsa1.perex.cz (Postfix) with ESMTP id 1E9DBF80129 for ; Wed, 12 May 2021 02:45:33 +0200 (CEST) DKIM-Filter: OpenDKIM Filter v2.11.0 alsa1.perex.cz 1E9DBF80129 Date: 12 May 2021 09:45:33 +0900 X-IronPort-AV: E=Sophos;i="5.82,291,1613401200"; d="scan'208";a="80799335" Received: from unknown (HELO relmlir5.idc.renesas.com) ([10.200.68.151]) by relmlie6.idc.renesas.com with ESMTP; 12 May 2021 09:45:33 +0900 Received: from mercury.renesas.com (unknown [10.166.252.133]) by relmlir5.idc.renesas.com (Postfix) with ESMTP id A18194016D44; Wed, 12 May 2021 09:45:33 +0900 (JST) Message-ID: <87y2ckaifm.wl-kuninori.morimoto.gx@renesas.com> From: Kuninori Morimoto Subject: [PATCH v2 2/7] ASoC: soc-core: add snd_soc_runtime_get_dai_fmt() User-Agent: Wanderlust/2.15.9 Emacs/26.3 Mule/6.0 MIME-Version: 1.0 (generated by SEMI-EPG 1.14.7 - "Harue") To: Mark Brown In-Reply-To: <871racbx0w.wl-kuninori.morimoto.gx@renesas.com> References: <871racbx0w.wl-kuninori.morimoto.gx@renesas.com> Cc: Linux-ALSA X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.15 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" From: Kuninori Morimoto ASoC is using dai_link which specify DAI format (= dai_link->dai_fmt), and it is selected by "Sound Card" driver in corrent implementation. In other words, Sound Card *needs* to setup it. But, it should be possible to automatically selected from CPU and Codec driver settings. This patch adds new snd_soc_runtime_get_dai_fmt() callback to help it. By this patch, dai_fmt can be automatically selected from each driver if both CPU / Codec driver had .get_dai_fmt callback. Automatically selectable *field* is depends on each drivers. For example, some driver want to select format "automatically", but want to select other fields "manually", because of complex limitation. Or other example, in case of both CPU and Codec are possible to be clock provider, but the quality was different. In these case, user need/want to *manually* select each fields from Sound Card driver. It uses Sound Card specified fields preferentially, and try to select non-specific fields from CPU and Codec driver settings if driver had callbacks. In other words, we can select all dai_fmt via Sound Card driver same as before. Link: https://lore.kernel.org/r/871rb3hypy.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Kuninori Morimoto --- v1 -> v2 - add more detail explanation on git-log, comment - don't specify dummy clock/frame settings. include/sound/soc-dai.h | 45 ++++++++++++++++ sound/soc/soc-core.c | 110 ++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-dai.c | 18 +++++++ sound/soc/soc-utils.c | 32 ++++++++++++ 4 files changed, 205 insertions(+) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 0bc29c4516e7..13ea51df283a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -36,6 +36,22 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J #define SND_SOC_DAIFMT_LSB SND_SOC_DAIFMT_RIGHT_J +/* Describes the possible PCM format */ +/* + * use SND_SOC_DAI_FORMAT_xx as eash shift. + * see + * snd_soc_runtime_get_dai_fmt() + */ +#define SND_SOC_POSSIBLE_DAIFMT_FORMAT_SHIFT 0 +#define SND_SOC_POSSIBLE_DAIFMT_FORMAT_MASK (0xFFFF << SND_SOC_POSSIBLE_DAIFMT_FORMAT_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_I2S (1 << SND_SOC_DAI_FORMAT_I2S) +#define SND_SOC_POSSIBLE_DAIFMT_RIGHT_J (1 << SND_SOC_DAI_FORMAT_RIGHT_J) +#define SND_SOC_POSSIBLE_DAIFMT_LEFT_J (1 << SND_SOC_DAI_FORMAT_LEFT_J) +#define SND_SOC_POSSIBLE_DAIFMT_DSP_A (1 << SND_SOC_DAI_FORMAT_DSP_A) +#define SND_SOC_POSSIBLE_DAIFMT_DSP_B (1 << SND_SOC_DAI_FORMAT_DSP_B) +#define SND_SOC_POSSIBLE_DAIFMT_AC97 (1 << SND_SOC_DAI_FORMAT_AC97) +#define SND_SOC_POSSIBLE_DAIFMT_PDM (1 << SND_SOC_DAI_FORMAT_PDM) + /* * DAI Clock gating. * @@ -45,6 +61,17 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ #define SND_SOC_DAIFMT_GATED (0 << 4) /* clock is gated */ +/* Describes the possible PCM format */ +/* + * define GATED -> CONT. GATED will be selected if both are selected. + * see + * snd_soc_runtime_get_dai_fmt() + */ +#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT 16 +#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_MASK (0xFFFF << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_GATED (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_CONT (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_SHIFT) + /* * DAI hardware signal polarity. * @@ -71,6 +98,14 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ #define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */ +/* Describes the possible PCM format */ +#define SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT 32 +#define SND_SOC_POSSIBLE_DAIFMT_INV_MASK (0xFFFFULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_NB_NF (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_NB_IF (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_IB_NF (0x4ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_IB_IF (0x8ULL << SND_SOC_POSSIBLE_DAIFMT_INV_SHIFT) + /* * DAI hardware clock providers/consumers * @@ -89,6 +124,14 @@ struct snd_compr_stream; #define SND_SOC_DAIFMT_CBM_CFS SND_SOC_DAIFMT_CBP_CFC #define SND_SOC_DAIFMT_CBS_CFS SND_SOC_DAIFMT_CBC_CFC +/* Describes the possible PCM format */ +#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT 48 +#define SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_MASK (0xFFFFULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_CBP_CFP (0x1ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_CBC_CFP (0x2ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_CBP_CFC (0x4ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) +#define SND_SOC_POSSIBLE_DAIFMT_CBC_CFC (0x8ULL << SND_SOC_POSSIBLE_DAIFMT_CLOCK_PROVIDER_SHIFT) + #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 #define SND_SOC_DAIFMT_INV_MASK 0x0f00 @@ -131,6 +174,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio); /* Digital Audio interface formatting */ +u64 snd_soc_dai_get_fmt(struct snd_soc_dai *dai); int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, @@ -236,6 +280,7 @@ struct snd_soc_dai_ops { * DAI format configuration * Called by soc_card drivers, normally in their hw_params. */ + u64 (*get_fmt)(struct snd_soc_dai *dai); int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*xlate_tdm_slot_mask)(unsigned int slots, unsigned int *tx_mask, unsigned int *rx_mask); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e241c35fb63a..194bf576b44d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1054,6 +1054,115 @@ int snd_soc_add_pcm_runtime(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_add_pcm_runtime); +static void snd_soc_runtime_get_dai_fmt(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai_link *dai_link = rtd->dai_link; + struct snd_soc_dai *dai; + u64 pos, possible_fmt = ULLONG_MAX; + unsigned int mask = 0, dai_fmt = 0; + int i; + + for_each_rtd_dais(rtd, i, dai) + possible_fmt &= snd_soc_dai_get_fmt(dai); + + if (!possible_fmt) + return; + + /* + * convert POSSIBLE_DAIFMT to DAIFMT + * + * Some basic/default settings on each is defined as 0. + * see + * SND_SOC_DAIFMT_NB_NF + * SND_SOC_DAIFMT_GATED + * + * SND_SOC_DAIFMT_xxx_MASK can't notice it if Sound Card specify + * these value, and will be overwrite to auto selected value. + * + * To avoid such issue, loop from 63 to 0 here. + * Small number of SND_SOC_POSSIBLE_xxx will be Hi priority. + * Basic/Default settings of each part and aboves are defined + * as Hi priority (= small number) of SND_SOC_POSSIBLE_xxx. + */ + for (i = 63; i >= 0; i--) { + pos = 1ULL << i; + switch (possible_fmt & pos) { + /* + * for format + */ + case SND_SOC_POSSIBLE_DAIFMT_I2S: + case SND_SOC_POSSIBLE_DAIFMT_RIGHT_J: + case SND_SOC_POSSIBLE_DAIFMT_LEFT_J: + case SND_SOC_POSSIBLE_DAIFMT_DSP_A: + case SND_SOC_POSSIBLE_DAIFMT_DSP_B: + case SND_SOC_POSSIBLE_DAIFMT_AC97: + case SND_SOC_POSSIBLE_DAIFMT_PDM: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_FORMAT_MASK) | i; + break; + /* + * for clock + */ + case SND_SOC_POSSIBLE_DAIFMT_CONT: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_CLOCK_MASK) | SND_SOC_DAIFMT_CONT; + break; + case SND_SOC_POSSIBLE_DAIFMT_GATED: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_CLOCK_MASK) | SND_SOC_DAIFMT_GATED; + break; + /* + * for clock invert + */ + case SND_SOC_POSSIBLE_DAIFMT_NB_NF: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_INV_MASK) | SND_SOC_DAIFMT_NB_NF; + break; + case SND_SOC_POSSIBLE_DAIFMT_NB_IF: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_INV_MASK) | SND_SOC_DAIFMT_NB_IF; + break; + case SND_SOC_POSSIBLE_DAIFMT_IB_NF: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_INV_MASK) | SND_SOC_DAIFMT_IB_NF; + break; + case SND_SOC_POSSIBLE_DAIFMT_IB_IF: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_INV_MASK) | SND_SOC_DAIFMT_IB_IF; + break; + /* + * for clock provider / consumer + */ + case SND_SOC_POSSIBLE_DAIFMT_CBP_CFP: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) | SND_SOC_DAIFMT_CBP_CFP; + break; + case SND_SOC_POSSIBLE_DAIFMT_CBC_CFP: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) | SND_SOC_DAIFMT_CBC_CFP; + break; + case SND_SOC_POSSIBLE_DAIFMT_CBP_CFC: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) | SND_SOC_DAIFMT_CBP_CFC; + break; + case SND_SOC_POSSIBLE_DAIFMT_CBC_CFC: + dai_fmt = (dai_fmt & ~SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) | SND_SOC_DAIFMT_CBC_CFC; + break; + } + } + + /* + * Some driver might have very complex limitation. + * In such case, user want to auto-select non-limitation part, + * and want to manually specify complex part. + * + * Or for example, if both CPU and Codec can be clock provider, + * but because of its quality, user want to specify it manually. + * + * Use manually specified settings if sound card did. + */ + if (!(dai_link->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK)) + mask |= SND_SOC_DAIFMT_FORMAT_MASK; + if (!(dai_link->dai_fmt & SND_SOC_DAIFMT_CLOCK_MASK)) + mask |= SND_SOC_DAIFMT_CLOCK_MASK; + if (!(dai_link->dai_fmt & SND_SOC_DAIFMT_INV_MASK)) + mask |= SND_SOC_DAIFMT_INV_MASK; + if (!(dai_link->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK)) + mask |= SND_SOC_DAIFMT_MASTER_MASK; + + dai_link->dai_fmt |= (dai_fmt & mask); +} + /** * snd_soc_runtime_set_dai_fmt() - Change DAI link format for a ASoC runtime * @rtd: The runtime for which the DAI link format should be changed @@ -1132,6 +1241,7 @@ static int soc_init_pcm_runtime(struct snd_soc_card *card, if (ret < 0) return ret; + snd_soc_runtime_get_dai_fmt(rtd); if (dai_link->dai_fmt) { ret = snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); if (ret) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 080fbe053fc5..762ae5251244 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -134,6 +134,24 @@ int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) } EXPORT_SYMBOL_GPL(snd_soc_dai_set_bclk_ratio); +/** + * snd_soc_dai_get_fmt - get enable DAI hardware audio format. + * @dai: DAI + * @fmt: SND_SOC_POSSIBLE_DAIFMT_* format value. + * + * Configures the DAI hardware format and clocking. + */ +u64 snd_soc_dai_get_fmt(struct snd_soc_dai *dai) +{ + u64 fmt = 0; + + if (dai->driver->ops && + dai->driver->ops->get_fmt) + fmt = dai->driver->ops->get_fmt(dai); + + return fmt; +} + /** * snd_soc_dai_set_fmt - configure DAI hardware audio format. * @dai: DAI diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 98383fd76224..ac53dc8c9a4c 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -97,6 +97,37 @@ static const struct snd_soc_component_driver dummy_codec = { SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) + +static u64 dummy_dai_get_fmt(struct snd_soc_dai *dai) +{ + /* + * dummy can be both CPU and Codec. + * Thus, it can't specify Clock/Frame provider/consumer here. + * (= SND_SOC_POSSIBLE_DAIFMT_CBP_CFP + * SND_SOC_POSSIBLE_DAIFMT_CBP_CFC + * SND_SOC_POSSIBLE_DAIFMT_CBC_CFP + * SND_SOC_POSSIBLE_DAIFMT_CBC_CFC) + */ + + return SND_SOC_POSSIBLE_DAIFMT_I2S | + SND_SOC_POSSIBLE_DAIFMT_RIGHT_J | + SND_SOC_POSSIBLE_DAIFMT_LEFT_J | + SND_SOC_POSSIBLE_DAIFMT_DSP_A | + SND_SOC_POSSIBLE_DAIFMT_DSP_B | + SND_SOC_POSSIBLE_DAIFMT_AC97 | + SND_SOC_POSSIBLE_DAIFMT_PDM | + SND_SOC_POSSIBLE_DAIFMT_GATED | + SND_SOC_POSSIBLE_DAIFMT_CONT | + SND_SOC_POSSIBLE_DAIFMT_NB_NF | + SND_SOC_POSSIBLE_DAIFMT_NB_IF | + SND_SOC_POSSIBLE_DAIFMT_IB_NF | + SND_SOC_POSSIBLE_DAIFMT_IB_IF; +} + +static const struct snd_soc_dai_ops dummy_dai_ops = { + .get_fmt = dummy_dai_get_fmt, +}; + /* * The dummy CODEC is only meant to be used in situations where there is no * actual hardware. @@ -122,6 +153,7 @@ static struct snd_soc_dai_driver dummy_dai = { .rates = STUB_RATES, .formats = STUB_FORMATS, }, + .ops = &dummy_dai_ops, }; int snd_soc_dai_is_dummy(struct snd_soc_dai *dai)