July 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 0/3] ASoC: codecs: msm8916-wcd-digital: Fix dmic support
by srinivas.kandagatla＠linaro.org 02 Aug '17

02 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patchset fixes dmic support in the codec driver, first and second patch fixes the mclk rate for the codec and the third patch adds missing selection of DMIC in CIC filter source path. Tested on DB410c with Audio Mezz board which has got dmic on it. Changes since v1: - check return values for set_sysclk and clk_set_rate() spoted by Mark. - Added more info in change log. - Fixed multicodec usecase. Srinivas Kandagatla (3): ASoC: codecs: msm8916-wcd-digital: add support to set_sysclk ASoC: qcom: apq8016-sbc: set default mclk rate ASoC: codecs: msm8916-wcd-digital: select DMIC source in CIC filter path sound/soc/codecs/msm8916-wcd-digital.c | 35 ++++++++++++++++++++++++++++++++++ sound/soc/qcom/apq8016_sbc.c | 23 ++++++++++++++++++---- 2 files changed, 54 insertions(+), 4 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH] streamline TLV320AIC23 drivers
by Jens Rottmann 01 Aug '17

01 Aug '17

The iMX-TLV320AIC23 driver isn't from Freescale, but from a company named Eukrea Electromatique, originally for their own boards. From the code I get the impression it is a bit older, its DT options use a differing naming scheme. Patch it up a bit: - Remove Eukrea naming, i.MX is from Freescale, TLV320AIC23 is from TI, driver was written by Eukrea, but it's DT capable, so it's not exclusive: - Kconfig option title - 'model' option - driver 'compatible' string - Other options just have changed over time, this driver remaining (one of) the last with the old semantics: - 'audio-codec' option (also moved from ssi node) - 'mux-int/ext-port' options - All options stay backwards compatible, the DT binding documents new and old names. CONFIG variable and files have not been renamed, though, so no need to change old defconfigs. Signed-off-by: Jens Rottmann <Jens.Rottmann(a)ADLINKtech.com> --- --- a/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt +++ b/Documentation/devicetree/bindings/sound/eukrea-tlv320.txt @@ -1,16 +1,23 @@ -Audio complex for Eukrea boards with tlv320aic23 codec. +Audio complex for Freescale i.MX boards with TI TLV320AIC23 I2S codecs, +like those from Eukrea Electromatique. Required properties: - - compatible : "eukrea,asoc-tlv320" + - compatible : "fsl,imx-audio-tlv320aic23" or + "eukrea,asoc-tlv320" (deprecated) - - eukrea,model : The user-visible name of this sound complex. + - model : The user-visible name of this sound complex. + - eukrea,model : Dito, deprecated. - ssi-controller : The phandle of the SSI controller. - - fsl,mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). + - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX). + - fsl,mux-int-port : Dito, deprecated. - - fsl,mux-ext-port : The external port of the i.MX audio muxer. + - mux-ext-port : The external port of the i.MX audio muxer. + - fsl,mux-ext-port : Dito, deprecated. + + - audio-codec : The phandle of the audio codec. Note: The AUDMUX port numbering should start at 1, which is consistent with hardware manual. @@ -18,9 +25,10 @@ hardware manual. Example: sound { - compatible = "eukrea,asoc-tlv320"; - eukrea,model = "imx51-eukrea-tlv320aic23"; + compatible = "fsl,imx-audio-tlv320aic23"; + model = "imx51-eukrea-tlv320aic23"; ssi-controller = <&ssi2>; - fsl,mux-int-port = <2>; - fsl,mux-ext-port = <3>; + mux-int-port = <2>; + mux-ext-port = <3>; + audio-codec = <&codec>; }; --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -220,7 +220,7 @@ config SND_SOC_PHYCORE_AC97 and phyCARD boards in AC97 mode config SND_SOC_EUKREA_TLV320 - tristate "Eukrea TLV320" + tristate "SoC Audio support for i.MX boards with TLV320AIC23" depends on ARCH_MXC && I2C select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -92,11 +92,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) eukrea_tlv320.dev = &pdev->dev; if (np) { - ret = snd_soc_of_parse_card_name(&eukrea_tlv320, - "eukrea,model"); + ret = snd_soc_of_parse_card_name(&eukrea_tlv320, "model"); + if (ret) /* backwards compatible */ + ret = snd_soc_of_parse_card_name(&eukrea_tlv320, + "eukrea,model"); if (ret) { dev_err(&pdev->dev, - "eukrea,model node missing or invalid.\n"); + "model node missing or invalid.\n"); goto err; } @@ -109,22 +111,28 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) goto err; } - codec_np = of_parse_phandle(ssi_np, "codec-handle", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!codec_np) /* backwards compatible */ + codec_np = of_parse_phandle(ssi_np, "codec-handle", 0); if (codec_np) eukrea_tlv320_dai.codec_of_node = codec_np; else - dev_err(&pdev->dev, "codec-handle node missing or invalid.\n"); + dev_err(&pdev->dev, "audio-codec node missing or invalid.\n"); - ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port); + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) /* backwards compatible */ + ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port); if (ret) { dev_err(&pdev->dev, - "fsl,mux-int-port node missing or invalid.\n"); + "mux-int-port node missing or invalid.\n"); return ret; } - ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) /* backwards compatible */ + ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, - "fsl,mux-ext-port node missing or invalid.\n"); + "mux-ext-port node missing or invalid.\n"); return ret; } @@ -213,7 +221,8 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) } static const struct of_device_id imx_tlv320_dt_ids[] = { - { .compatible = "eukrea,asoc-tlv320"}, + { .compatible = "eukrea,asoc-tlv320"}, /* backwards compatible */ + { .compatible = "fsl,imx-audio-tlv320aic23"}, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); _

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[alsa-devel] [PATCH v2 0/2] ASoC: hdmi-codec: Allow driver to restrict sample size
by srinivas.kandagatla＠linaro.org 01 Aug '17

01 Aug '17

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> This patch adds option to allow hdmi-codec drivers to restrict the audio sample size based on the values that are suppored in hardware. First patch is adding that formats option to hdmi_codec_pdata and second patch is a fix in adv7511 codec driver which only supports 16 and 24 bit samples. Changes since v1: - Added Review by Jyri - Updated comment in code suggested by Jyri Srinivas Kandagatla (2): ASoC: hdmi-codec: Allow drivers to restrict sample sizes. drm/bridge: adv7511: restrict audio sample sizes drivers/gpu/drm/bridge/adv7511/adv7511_audio.c | 2 ++ include/sound/hdmi-codec.h | 1 + sound/soc/codecs/hdmi-codec.c | 8 +++++--- 3 files changed, 8 insertions(+), 3 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH] ASoC: codecs: fix wm8524 build error
by Arnd Bergmann 01 Aug '17

01 Aug '17

When CONFIG_OF is disabled, we run into a build error: sound/soc/codecs/wm8524.c:257:21: error: 'wm8524_of_match' undeclared here (not in a function); did you mean 'wm8524_dai'? This removes the unnecessary #ifdef around the match table. Fixes: 007b6a54c305 ("ASoC: codecs: add wm8524 codec driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/wm8524.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index f42ec664877c..e22b76d64e32 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -206,13 +206,11 @@ static const struct snd_soc_codec_driver soc_codec_dev_wm8524 = { }, }; -#ifdef CONFIG_OF static const struct of_device_id wm8524_of_match[] = { { .compatible = "wlf,wm8524" }, { /* sentinel*/ } }; MODULE_DEVICE_TABLE(of, wm8524_of_match); -#endif static int wm8524_codec_probe(struct platform_device *pdev) { -- 2.9.0

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[alsa-devel] [PATCH 1/4] ASoC: Intel: Skylake: Reset the controller in probe
by Guneshwor Singh 01 Aug '17

01 Aug '17

Controller can be in reset state by default. Capability structure traversal requires the controller to be out of reset else it results in broken capability parsing. Hence make sure that controller is out of reset before parsing capabilities by doing a full reset. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/skylake/skl.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 334917ee41cf..a56dfde3928c 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -702,6 +702,8 @@ static int skl_first_init(struct hdac_ext_bus *ebus) return -ENXIO; } + skl_init_chip(bus, true); + snd_hdac_bus_parse_capabilities(bus); if (skl_acquire_irq(ebus, 0) < 0) -- 2.13.0

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[alsa-devel] [PATCH 1/2] ASoC: rt5514: Add the sanity checks of the buffer related address
by Oder Chiou 01 Aug '17

01 Aug '17

The patch add the sanity checks of the buffer related address to make sure the addresses are valid. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 950d1ff..6f513cc 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -173,9 +173,14 @@ static irqreturn_t rt5514_spi_irq(int irq, void *data) rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | buf[3] << 24; + if (rt5514_dsp->buf_rp % 8) + rt5514_dsp->buf_rp = (rt5514_dsp->buf_rp / 8) * 8; + rt5514_dsp->buf_size = rt5514_dsp->buf_limit - rt5514_dsp->buf_base; - schedule_delayed_work(&rt5514_dsp->copy_work, 0); + if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && + rt5514_dsp->buf_rp && rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 0); return IRQ_HANDLED; } -- 2.7.4

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[alsa-devel] [PATCH] device property: Fix usecount for of_graph_get_port_parent()
by Tony Lindgren 01 Aug '17

01 Aug '17

Fix inconsistent use of of_graph_get_port_parent() where asoc_simple_card_parse_graph_dai() does of_node_get() before calling it while other callers do not. We can fix this by not trashing the node passed to of_graph_get_port_parent(). Let's make sure the users have correct refcounts and remove related incorrect of_node_put() calls for of_for_each_phandle as that's done by of_phandle_iterator_next(). Let's fix both issues with a single patch to avoid kobject refcounts getting messed up more if two patches are merged separately. Otherwise strange issues can happen caused by memory corruption caused by too many kobject_del() calls such as: BUG: sleeping function called from invalid context at kernel/locking/mutex.c:747 ... (___might_sleep) (__mutex_lock) (mutex_lock_nested) (kernfs_remove) (kobject_del) (kobject_put) (of_get_next_parent) (of_graph_get_port_parent) (asoc_simple_card_parse_graph_dai [snd_soc_simple_card_utils]) (asoc_graph_card_probe [snd_soc_audio_graph_card]) Fixes: 0ef472a973eb ("of_graph: add of_graph_get_port_parent()") Fixes: 2692c1c63c29 ("ASoC: add audio-graph-card support") Fixes: 1689333f8311 ("ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai()") Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: alsa-devel(a)alsa-project.org Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- drivers/of/property.c | 9 +++++++++ sound/soc/generic/audio-graph-card.c | 5 +---- sound/soc/generic/simple-card-utils.c | 5 ----- 3 files changed, 10 insertions(+), 9 deletions(-) diff --git a/drivers/of/property.c b/drivers/of/property.c --- a/drivers/of/property.c +++ b/drivers/of/property.c @@ -708,6 +708,15 @@ struct device_node *of_graph_get_port_parent(struct device_node *node) { unsigned int depth; + if (!node) + return NULL; + + /* + * Preserve usecount for passed in node as of_get_next_parent() + * will do of_node_put() on it. + */ + of_node_get(node); + /* Walk 3 levels up only if there is 'ports' node. */ for (depth = 3; depth && node; depth--) { node = of_get_next_parent(node); diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -224,7 +224,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { ret = asoc_graph_card_dai_link_of(it.node, priv, idx++); - of_node_put(it.node); if (ret < 0) return ret; } @@ -239,10 +238,8 @@ static int asoc_graph_get_dais_count(struct device *dev) int count = 0; int rc; - of_for_each_phandle(&it, rc, node, "dais", NULL, 0) { + of_for_each_phandle(&it, rc, node, "dais", NULL, 0) count++; - of_node_put(it.node); - } return count; } diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -282,11 +282,6 @@ int asoc_simple_card_parse_graph_dai(struct device_node *ep, if (!dai_name) return 0; - /* - * of_graph_get_port_parent() will call - * of_node_put(). So, call of_node_get() here - */ - of_node_get(ep); node = of_graph_get_port_parent(ep); /* Get dai->name */ -- 2.13.2

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[alsa-devel] [PATCH] ASoC: max98090 - Expose HP out enable.
by Enric Balletbo i Serra 01 Aug '17

01 Aug '17

From: Hsin-Yu Chao <hychao(a)chromium.org> Expose the headphone output enables to user space. This will allow userspace to ensure that there is stable output before the headphones are turned on, avoiding a pop on insertion. Signed-off-by: Hsin-Yu Chao <hychao(a)chromium.org> Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- sound/soc/codecs/max98090.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6682848..f38c051 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1074,6 +1074,15 @@ static SOC_ENUM_SINGLE_DECL(mixhprsel_mux_enum, static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); +/* HP output enables. */ +static const struct snd_kcontrol_new max98090_hpl_enable = + SOC_DAPM_SINGLE("Switch", M98090_REG_OUTPUT_ENABLE, + M98090_HPLEN_SHIFT, 1, 0); + +static const struct snd_kcontrol_new max98090_hpr_enable = + SOC_DAPM_SINGLE("Switch", M98090_REG_OUTPUT_ENABLE, + M98090_HPREN_SHIFT, 1, 0); + static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), @@ -1218,10 +1227,10 @@ static const struct snd_soc_dapm_widget max98090_dapm_widgets[] = { SND_SOC_DAPM_MUX("MIXHPRSEL Mux", M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT, 0, &max98090_mixhprsel_mux), - SND_SOC_DAPM_PGA("HP Left Out", M98090_REG_OUTPUT_ENABLE, - M98090_HPLEN_SHIFT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP Right Out", M98090_REG_OUTPUT_ENABLE, - M98090_HPREN_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("HP Left Out", SND_SOC_NOPM, 0, 0, + &max98090_hpl_enable), + SND_SOC_DAPM_SWITCH("HP Right Out", SND_SOC_NOPM, 0, 0, + &max98090_hpr_enable), SND_SOC_DAPM_PGA("SPK Left Out", M98090_REG_OUTPUT_ENABLE, M98090_SPLEN_SHIFT, 0, NULL, 0), @@ -1405,8 +1414,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { * Disable this for lowest power if bypassing * the DAC with an analog signal */ - {"HP Left Out", NULL, "DACL"}, - {"HP Left Out", NULL, "MIXHPLSEL Mux"}, + {"HP Left Out", "Switch", "DACL"}, + {"HP Left Out", "Switch", "MIXHPLSEL Mux"}, {"MIXHPRSEL Mux", "HP Mixer", "Right Headphone Mixer"}, @@ -1414,8 +1423,8 @@ static const struct snd_soc_dapm_route max98090_dapm_routes[] = { * Disable this for lowest power if bypassing * the DAC with an analog signal */ - {"HP Right Out", NULL, "DACR"}, - {"HP Right Out", NULL, "MIXHPRSEL Mux"}, + {"HP Right Out", "Switch", "DACR"}, + {"HP Right Out", "Switch", "MIXHPRSEL Mux"}, {"SPK Left Out", NULL, "Left Speaker Mixer"}, {"SPK Right Out", NULL, "Right Speaker Mixer"}, -- 2.9.3

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[alsa-devel] [PATCH] davinci-evm: extend it to support DM81xx and derivatives
by Vitaly Wool 31 Jul '17

31 Jul '17

Having worked closely with TI DM8148/DM8168 EVMs I came across many derivatives of these boards which didn't use the standard codec but had something like wm897x instead. They all would fit davinci-evm driver nicely if it had support for audio CPU clock divisor and slightly more generic naming. So this patch adds support for reading CPU divisor from a DT and slightly changes names of helper/init functions to be more generic. It also adds support for DaVinci boards that use wm8978 codec. Signed-off-by: Vitaly Wool <vitaly.wool(a)konsulko.com> --- .../bindings/sound/davinci-evm-audio.txt | 6 +- sound/soc/davinci/davinci-evm.c | 66 ++++++++++++++++++++-- 2 files changed, 65 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt index 963e100514c2..e76966a06f22 100644 --- a/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-evm-audio.txt @@ -1,7 +1,9 @@ -* Texas Instruments SoC audio setups with TLV320AIC3X Codec +* Texas Instruments SoC audio setups with TLV320AIC3X/WM89XX codecs Required properties: - compatible : "ti,da830-evm-audio" : forDM365/DA8xx/OMAPL1x/AM33xx + "ti,dm8168-evm-audio" : for DM81xx with TLV320AIC3x + "ti,dm8168-evm-audio-wm8978" : for DM81xx with WM897x - ti,model : The user-visible name of this sound complex. - ti,audio-codec : The phandle of the TLV320AIC3x audio codec - ti,mcasp-controller : The phandle of the McASP controller @@ -14,6 +16,8 @@ Optional properties: - ti,codec-clock-rate : The Codec Clock rate (in Hz) applied to the Codec. - clocks : Reference to the master clock - clock-names : The clock should be named "mclk" +- ti,cpu-clkdiv : CPU clock divisor (if codec and cpu are clocked at + different rates) - Either codec-clock-rate or the codec-clock reference has to be defined. If the both are defined the driver attempts to set referenced clock to the defined rate and takes the rate from the clock reference. diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 7a369e0f2093..aa8e11583a84 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -26,7 +26,8 @@ struct snd_soc_card_drvdata_davinci { struct clk *mclk; - unsigned sysclk; + unsigned int sysclk; + unsigned int clkdiv; }; static int evm_startup(struct snd_pcm_substream *substream) @@ -61,8 +62,11 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_card *soc_card = rtd->card; int ret = 0; - unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) - snd_soc_card_get_drvdata(soc_card))->sysclk; + unsigned int sysclk = ((struct snd_soc_card_drvdata_davinci *) + snd_soc_card_get_drvdata(soc_card))->sysclk; + unsigned int clkdiv = ((struct snd_soc_card_drvdata_davinci *) + snd_soc_card_get_drvdata(soc_card))->clkdiv; + /* set the codec system clock */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT); @@ -74,6 +78,13 @@ static int evm_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set the CPU system clock divisor */ + if (clkdiv) { + ret = snd_soc_dai_set_clkdiv(cpu_dai, 0, clkdiv); + if (ret < 0) + return ret; + } + return 0; } @@ -84,7 +95,7 @@ static struct snd_soc_ops evm_ops = { }; /* davinci-evm machine dapm widgets */ -static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { +static const struct snd_soc_dapm_widget evm_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_LINE("Line Out", NULL), SND_SOC_DAPM_MIC("Mic Jack", NULL), @@ -121,8 +132,8 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) int ret; /* Add davinci-evm specific widgets */ - snd_soc_dapm_new_controls(&card->dapm, aic3x_dapm_widgets, - ARRAY_SIZE(aic3x_dapm_widgets)); + snd_soc_dapm_new_controls(&card->dapm, evm_dapm_widgets, + ARRAY_SIZE(evm_dapm_widgets)); if (np) { ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); @@ -142,6 +153,28 @@ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) return 0; } +/* Logic for a non-aic3x (e. g. wm89xx) connected to a davinci-evm */ +static int evm_simple_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + struct device_node *np = card->dev->of_node; + + /* Add davinci-evm specific widgets */ + snd_soc_dapm_new_controls(&card->dapm, evm_dapm_widgets, + ARRAY_SIZE(evm_dapm_widgets)); + + /* audio routing is optional here */ + if (np) + snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); + + /* not connected */ + snd_soc_dapm_nc_pin(&card->dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(&card->dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(&card->dapm, "HPRCOM"); + + return 0; +} + /* davinci-evm digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link dm6446_evm_dai = { .name = "TLV320AIC3X", @@ -340,11 +373,29 @@ static struct snd_soc_dai_link evm_dai_tlv320aic3x = { SND_SOC_DAIFMT_IB_NF, }; +static struct snd_soc_dai_link evm_dai_wm8978 = { + .name = "wm8978", + .stream_name = "wm8978", + .codec_dai_name = "wm8978-hifi", + .ops = &evm_ops, + .init = evm_simple_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_IB_NF, +}; + static const struct of_device_id davinci_evm_dt_ids[] = { { .compatible = "ti,da830-evm-audio", .data = (void *) &evm_dai_tlv320aic3x, }, + { + .compatible = "ti,dm8168-evm-audio", + .data = (void *) &evm_dai_tlv320aic3x, + }, + { + .compatible = "ti,dm8168-evm-audio-wm8978", + .data = (void *) &evm_dai_wm8978, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, davinci_evm_dt_ids); @@ -422,6 +473,9 @@ static int davinci_evm_probe(struct platform_device *pdev) requestd_rate, drvdata->sysclk); } + /* CPU clock divisor may need to be provided for e. g. wm897x */ + of_property_read_u32(np, "ti,cpu-clkdiv", &drvdata->clkdiv); + snd_soc_card_set_drvdata(&evm_soc_card, drvdata); ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card); -- 2.11.0

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[alsa-devel] [PATCH] ASoC: audio-graph-scu-card: fix balance of of_node_get/put()
by Kuninori Morimoto 31 Jul '17

31 Jul '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> audio-graph-scu-card with CONFIG_OF_DYNAMIC triggers several errors at kernel boot, like the following one OF: ERROR: Bad of_node_put() on /soc/sound@ec500000/ports/port@1 This patch doesn't call unnecessary of_node_put() which was called in other function (= of_graph_get_port_parent(), of_phandle_iterator_next()) Reported-by: Antonio Borneo <borneo.antonio(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-scu-card.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/generic/audio-graph-scu-card.c b/sound/soc/generic/audio-graph-scu-card.c index dcd2df3..bff1e6a 100644 --- a/sound/soc/generic/audio-graph-scu-card.c +++ b/sound/soc/generic/audio-graph-scu-card.c @@ -215,7 +215,6 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) codec_ep = of_graph_get_remote_endpoint(cpu_ep); rcpu_ep = of_graph_get_remote_endpoint(codec_ep); - of_node_put(cpu_port); of_node_put(cpu_ep); of_node_put(codec_ep); of_node_put(rcpu_ep); @@ -248,11 +247,9 @@ static int asoc_graph_card_parse_of(struct graph_card_data *priv) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); + codec_port = of_graph_get_port_parent(codec_ep); /* call put */ - of_node_put(cpu_port); of_node_put(cpu_ep); - of_node_put(codec_ep); of_node_put(codec_port); if (codec) { @@ -304,11 +301,9 @@ static int asoc_graph_get_dais_count(struct device *dev) cpu_port = it.node; cpu_ep = of_get_next_child(cpu_port, NULL); codec_ep = of_graph_get_remote_endpoint(cpu_ep); - codec_port = of_graph_get_port_parent(codec_ep); + codec_port = of_graph_get_port_parent(codec_ep); /* call put */ - of_node_put(cpu_port); of_node_put(cpu_ep); - of_node_put(codec_ep); of_node_put(codec_port); if (cpu_ep) -- 1.9.1

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