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

[alsa-devel] [PATCH RESEND v3 0/3] Add support for es8323 and hdmi audio to the firefly-rk3288
by Romain Perier 04 Aug '17

04 Aug '17

This set of patches enables support for the es8323 codecs on the firefly-rk3288. It adds the required DT definitions to link rockchip-i2s to the es8388 analog output. As the support for HDMI audio is also built-in in the corresponding ASoC machine driver, HDMI audio output is also enabled. Changes in v3: - Synced DT with the one of the Rock2, the properties have been lexically re-ordered in the right order, I have splitted the headphone pinctrl in two: phone_ctl and hp_det Changes in v2: - Adapted DT for the firefly-rk3288 to the ASoC machine driver rk3288-hdmi-analog - Added patch 02/03 to this series - Re-wrote this cover letter Romain Perier (3): SoC: es8328-i2c: Add compatible for ES8323 arm: dts: rockchip: Don't set otp-gpio pinctrl by default in rk3288.dtsi arm: dts: Add support for ES8323 to the Firefly-RK3288 Documentation/devicetree/bindings/sound/es8328.txt | 5 ++- arch/arm/boot/dts/rk3288-evb.dtsi | 4 +++ arch/arm/boot/dts/rk3288-firefly.dtsi | 38 ++++++++++++++++++++++ arch/arm/boot/dts/rk3288-veyron.dtsi | 4 +++ arch/arm/boot/dts/rk3288.dtsi | 4 --- sound/soc/codecs/es8328-i2c.c | 2 ++ 6 files changed, 52 insertions(+), 5 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH 0/2] ASoC: codec: add DT support to dmic codec
by Arnaud Pouliquen 02 Aug '17

02 Aug '17

These patches add of_table description in dmic codec to support DT probing. As this codec is a generic code to support simple digital microphone, compatibility field is generic. Arnaud Pouliquen (2): ASoC: Add bindings for DMIC codec driver ASoC: codec: add DT support in dmic codec Documentation/devicetree/bindings/sound/dmic.txt | 12 ++++++++++++ sound/soc/codecs/dmic.c | 6 ++++++ 2 files changed, 18 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/dmic.txt -- 2.7.4

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[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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