February 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] sound/tlv320dac33: Add device tree support
by Pavel Machek 05 Mar '18

05 Mar '18

This adds device tree support to tlv320dac33.c. Signed-off-by: Pavel Machek <pavel(a)ucw.cz> diff --git a/Documentation/devicetree/bindings/sound/tlv320dac33.txt b/Documentation/devicetree/bindings/sound/tlv320dac33.txt new file mode 100644 index 0000000..6cbd311 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320dac33.txt @@ -0,0 +1,32 @@ +Texas Instruments - tlv320dac33 Codec module + +The tlv320dac33 serial control bus communicates through I2C protocols. + +Required properties: + +- compatible - "ti,tlv320dac33" +- reg - I2C slave address + +Optional properties: + +- power-gpios - gpio pin to power the device, active high + +- interrupts - The interrupt output from the device. +- interrupt-parent - The parent interrupt controller. + +- ti,keep-bclk - Keep the BCLK running in FIFO modes +- ti,burst-bclkdiv - BCLK divider value in burst mode + +Example: + +tlv320dac33: audio-codec@19 { + compatible = "ti,tlv320dac33"; + reg = <0x19>; + + interrupt-parent = <&gpio2>; + interrupts = <21 1>; /* gpio_53, IRQF_TRIGGER_RISING */ + power-gpio = <&gpio2 28 0>; /* gpio_60 */ + + ti,keep-bclk; + ti,burst-bclkdiv = <3>; +}; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 5b94a15..2a0fe06 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +32,7 @@ #include <linux/interrupt.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -105,7 +108,7 @@ struct tlv320dac33_priv { unsigned int nsample; /* burst read amount from host */ int mode1_latency; /* latency caused by the i2c writes in * us */ - u8 burst_bclkdiv; /* BCLK divider value in burst mode */ + u32 burst_bclkdiv; /* BCLK divider value in burst mode */ unsigned int burst_rate; /* Interface speed in Burst modes */ int keep_bclk; /* Keep the BCLK continuously running @@ -1484,16 +1487,11 @@ static struct snd_soc_dai_driver dac33_dai = { static int dac33_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_platform_data *pdata = client->dev.platform_data; struct tlv320dac33_priv *dac33; + struct device_node *np = client->dev.of_node; int ret, i; - if (client->dev.platform_data == NULL) { - dev_err(&client->dev, "Platform data not set\n"); - return -ENODEV; - } - pdata = client->dev.platform_data; - dac33 = devm_kzalloc(&client->dev, sizeof(struct tlv320dac33_priv), GFP_KERNEL); if (dac33 == NULL) @@ -1505,10 +1503,26 @@ static int dac33_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, dac33); - dac33->power_gpio = pdata->power_gpio; - dac33->burst_bclkdiv = pdata->burst_bclkdiv; - dac33->keep_bclk = pdata->keep_bclk; - dac33->mode1_latency = pdata->mode1_latency; + if (pdata) { + dac33->power_gpio = pdata->power_gpio; + dac33->burst_bclkdiv = pdata->burst_bclkdiv; + dac33->keep_bclk = pdata->keep_bclk; + dac33->mode1_latency = pdata->mode1_latency; + } else if (np) { + ret = of_get_named_gpio(np, "power-gpios", 0); + if (ret >= 0) + dac33->power_gpio = ret; + else + dac33->power_gpio = -1; + + if (of_property_read_bool(np, "ti,keep-bclk")) + dac33->keep_bclk = true; + + of_property_read_u32(np, "ti,burst-bclkdiv", &dac33->burst_bclkdiv); + } else { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } if (!dac33->mode1_latency) dac33->mode1_latency = 10000; /* 10ms */ dac33->irq = client->irq; @@ -1574,9 +1588,16 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); +static const struct of_device_id tlv320dac33_of_match[] = { + { .compatible = "ti,tlv320dac33", }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, tlv320dac33_of_match); + static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { .name = "tlv320dac33-codec", + .of_match_table = of_match_ptr(tlv320dac33_of_match), }, .probe = dac33_i2c_probe, .remove = dac33_i2c_remove, -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

5 17

[alsa-devel] regression v4.16 on Nokia N900:/dev/input/event6 aka AV Jack support disappeared
by Pavel Machek 02 Mar '18

02 Mar '18

Hi! In v4.16, AV jack support disappeared. This one is suspect: commit 7be4b5dc7ffa9499ac6ef33a5ffa9ff43f9b7057 Author: Andrew F. Davis <afd(a)ti.com> Date: Wed Nov 29 11:13:59 2017 -0600 ARM: dts: omap3-n900: Fix the audio CODEC's reset pin The correct DT property for specifying a GPIO used for reset is "reset-gpios", fix this here. Fixes: 14e3e295b2b9 ("ARM: dts: omap3-n900: Add TLV320AIC3X support") Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Tony Lindgren <tony(a)atomide.com> How was it tested? It reverts polarity of reset pin, but sound/soc/codecs/tlv320aic3x.c treats those as aliases: ret = of_get_named_gpio(np, "reset-gpios", 0); if (ret >= 0) { aic3x->gpio_reset = ret; } else { ret = of_get_named_gpio(np, "gpio-reset", 0); if (ret > 0) { dev_warn(&i2c->dev, "Using deprecated property \"gpio-r\eset\", please update your DT"); aic3x->gpio_reset = ret; Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

6 19

[alsa-devel] [PATCH 0/5] ASoC: Intel: Skylake fixes
by Guneshwor Singh 01 Mar '18

01 Mar '18

This includes fixes for Skylake+ driver which are improving widget type check, fixes for simultaneous wake-on-voice (WoV) and capture use-case, fix to improve firmware download latency and lastly, fix to use uuid instead of module id for library modules. Guneshwor Singh (1): ASoC: Intel: Skylake: Refine skl widget type check Pawse, GuruprasadX (2): ASoC: Intel: Skylake: Add BE DAI for WoV usecase ASoC: Intel: board: Add BE DAI link for WoV and update DAPM machine map. Sanyog Kale (1): ASoC: Intel: Skylake: Disable clock and power gating during fw/lib download Sriram Periyasamy (1): ASoC: Intel: Skylake: Find module id from UUID for bind params sound/soc/intel/boards/bxt_rt298.c | 15 +++++ sound/soc/intel/skylake/skl-messages.c | 7 ++- sound/soc/intel/skylake/skl-pcm.c | 20 +++++- sound/soc/intel/skylake/skl-sst-ipc.h | 3 + sound/soc/intel/skylake/skl-topology.c | 107 +++++++++++++++++++++++++++++---- sound/soc/intel/skylake/skl-topology.h | 13 +++- sound/soc/intel/skylake/skl.c | 21 +++++++ sound/soc/intel/skylake/skl.h | 4 ++ 8 files changed, 176 insertions(+), 14 deletions(-) -- 2.16.1

2 10

[alsa-devel] xhci: Not enough bandwidth for new device state
by Waldemar Brodkorb 01 Mar '18

01 Mar '18

Hi, I am getting these messages from a Debian 9.3 system using Linux 4.9.x: [ 547.352746] usb 3-5.2: 3:1: usb_set_interface failed (-28) [ 548.352865] usb 3-5.3: Not enough bandwidth for new device state. [ 548.352868] usb 3-5.3: Not enough bandwidth for altsetting 1 I have connected a 7-Port USB 3.0 Hub to my machine. I have 6 different Headsets and USB Soundcards connected to the HUB. As soon as I start a C++ application which opens all ALSA devices I get these errors and the USB stopps working correctly. I think the maximum count of USB sound devices, which behave stable is three or four. Is this some miscalculation of the Linux kernel regarding the USB bandwidth usage? How I can determine how much bandwidth the USB sound card driver preallocates? Or are five USB soundcards really too much for an USB 3.0 bus? Thanks in advance for any advise, best regards Waldemar P.s.: I sent this message to LKML, but maybe it is more appropriate to ask here.

2 3

[alsa-devel] [PATCH] ASoC: Intel: kbl: Enable mclk and ssp sclk early for rt5663
by Naveen Manohar 01 Mar '18

01 Mar '18

rt5663 requires mclk/sclk early to synchronize its internal clocks. Enable these clocks early. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Kevin Cheng <kevin.cheng(a)intel.com> --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 3c9585eb6860..0e6b7e79441c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -204,13 +204,18 @@ static const struct snd_soc_dapm_widget kabylake_5663_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_SPK("DP", NULL), SND_SOC_DAPM_SPK("HDMI", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_5663_map[] = { + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, -- 2.7.4

2 1

[alsa-devel] [RFC 0/7] ASoC: rt5651: Drop pdata, move over to device-properties
by Hans de Goede 01 Mar '18

01 Mar '18

Hi Mark, et al., Here is a second attempt to stop having duplicated DMI quirk entries between the rt5651 codec and machine drivers. Note this is only compile tested, I'm looking for some quick feedback if this seems to be going in the right direction, as I'm planning on rebuilding the rest of my jack-detect cleanup / fix series on top of this. The diffstat for this entire set is: Documentation/devicetree/bindings/sound/rt5651.txt | 6 include/sound/rt5651.h | 12 sound/soc/codecs/rt5651.c | 375 +++++++++------------ sound/soc/codecs/rt5651.h | 7 sound/soc/intel/boards/bytcr_rt5651.c | 46 ++ 5 files changed, 213 insertions(+), 233 deletions(-) So besides making the code cleaner (IMHO) it also makes it somewhat smaller and as an added bonus it fixes support the DMIC_EN quirk which was broken. Regards, Hans

2 9

[alsa-devel] [PATCH 0/7] Let Atmel use simple-audio-card
by Ladislav Michl 01 Mar '18

01 Mar '18

Alexandre, following patchset is sent as a bare minimum I need to use MAX9867 codec in master mode on AT91SAM9G20 based board. Codec's MCLK is driven by PCK0 and SSC is wired as follows: RK0 <-- BCLK RF0 <-- LRCLK TD0 --> SDIN RD0 <-- SDOUT Following DT framgment was used: i2c-gpio-0 { codec: max9867@18 { compatible = "maxim,max9867"; reg = <0x18>; #sound-dai-cells = <0>; }; }; sound { compatible = "simple-audio-card"; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_pck0_as_mck>; simple-audio-card,name = "simear4-max9867"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&codec_dai>; simple-audio-card,frame-master = <&codec_dai>; simple-audio-card,routing = "LINE_IN", "Line In Jack", "Speaker", "HPOUT"; simple-audio-card,widgets = "Line", "Line In Jack", "Speaker", "Speaker"; cpu_dai: simple-audio-card,cpu { sound-dai = <&ssc0>; }; codec_dai: simple-audio-card,codec { sound-dai = <&codec>; clocks = <&pck0>; clock-frequency = <12384000>; }; }; &ssc0 { status = "okay"; pinctrl-0 = <&pinctrl_board_ssc0_tx &pinctrl_ssc0_rx>; atmel,clk-from-rk-pin; #sound-dai-cells = <0>; }; Codec code itself seems to be unused since it was merged and I'm wondering whenever it worked at all. A comment from codec author would help a lot, just to be sure I did not break anything. By no means this code is ready to be merged (only except those patches with commit log and after carefull consideration :)), I'm posting it as a base for discussion. Thank you, ladis Ladislav Michl (7): ASoC: atmel: Remove redundant dev_err() call in probe function ASoC: atmel_ssc_dai: Fix TCMR settings in I2S slave mode ASoC: simple_card_utils: Set clock frequency ASoC: max9867: Show Kconfig entry ASoC: max9867: Calculate LRCLK divider ASoC: max9867: Fix BSEL value in master mode. ASoC: max9867: Take chip out of shutdown sound/soc/atmel/atmel-classd.c | 7 +-- sound/soc/atmel/atmel-pdmic.c | 7 +-- sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/max9867.c | 110 +++------------------------------- sound/soc/generic/simple-card-utils.c | 4 ++ 6 files changed, 19 insertions(+), 114 deletions(-) -- 2.15.1

4 19

[alsa-devel] [PATCH v1 0/4] ASoC: Add support for DAC PCM1789
by Mylène Josserand 01 Mar '18

01 Mar '18

Hello everyone, You will find in this series the support of Texas Instrument's DAC PCM1789. This DAC is very minimalist and is similar to PCM1792a except for some differences in registers. Series based on asoc tree, "for-next" branch (last commit 130c3888dfdc). It is important to notice that this DAC needs to always have clocks enabled (even without any data) otherwise it will be in a "desynchronized" state and can not send data correctly. This issue has been solved by performing a reset each time a sound is played (see patch 04). This reset can produce a "pop" noise. Thank you in advance for any review. Best regards, Mylène Mylène Josserand (4): ASoC: codecs: pcm179x: Add PCM1789 id ASoC: codecs: pcm179x: Add support for PCM1789 ASoC: codecs: pcm179x: Add reset gpio ASoC: codecs: pcm179x: Add trigger function to perform a reset .../devicetree/bindings/sound/pcm179x.txt | 2 +- sound/soc/codecs/pcm179x-i2c.c | 17 +- sound/soc/codecs/pcm179x.c | 242 ++++++++++++++++++++- sound/soc/codecs/pcm179x.h | 10 +- 4 files changed, 264 insertions(+), 7 deletions(-) -- 2.11.0

4 14

[alsa-devel] [PATCH v4] ASoC: imx-wm8962: Remove machine driver
by Fabio Estevam 28 Feb '18

28 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> fsl-asoc-card machine driver also handles wm8962, so there is really no need for keeping the dedicated imx-wm8962 driver anymore. Remove the imx-wm8962 machine driver. Suggested-by: Nicolin Chen <nicoleotsuka(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v3: - Remove the imx-audio-wm8962.txt reference from fsl-asoc-card.txt. (Nicolin) .../devicetree/bindings/sound/fsl-asoc-card.txt | 1 - .../devicetree/bindings/sound/imx-audio-wm8962.txt | 53 ---- sound/soc/fsl/Kconfig | 11 - sound/soc/fsl/Makefile | 2 - sound/soc/fsl/imx-wm8962.c | 312 --------------------- 5 files changed, 379 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt delete mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index f749e27..c60a573 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -28,7 +28,6 @@ The compatible list for this generic sound card currently: (compatible with CS4271 and CS4272) "fsl,imx-audio-wm8962" - (compatible with Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt) "fsl,imx-audio-sgtl5000" (compatible with Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt) diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt deleted file mode 100644 index acea71b..0000000 --- a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt +++ /dev/null @@ -1,53 +0,0 @@ -Freescale i.MX audio complex with WM8962 codec - -Required properties: - - - compatible : "fsl,imx-audio-wm8962" - - - model : The user-visible name of this sound complex - - - ssi-controller : The phandle of the i.MX SSI controller - - - audio-codec : The phandle of the WM8962 audio codec - - - audio-routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names could be power supplies, WM8962 - pins, and the jacks on the board: - - Power supplies: - * Mic Bias - - Board connectors: - * Mic Jack - * Headphone Jack - * Ext Spk - - - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) - - - mux-ext-port : The external port of the i.MX audio muxer - -Note: The AUDMUX port numbering should start at 1, which is consistent with -hardware manual. - -Example: - -sound { - compatible = "fsl,imx6q-sabresd-wm8962", - "fsl,imx-audio-wm8962"; - model = "wm8962-audio"; - ssi-controller = <&ssi2>; - audio-codec = <&codec>; - audio-routing = - "Headphone Jack", "HPOUTL", - "Headphone Jack", "HPOUTR", - "Ext Spk", "SPKOUTL", - "Ext Spk", "SPKOUTR", - "MICBIAS", "AMIC", - "IN3R", "MICBIAS", - "DMIC", "MICBIAS", - "DMICDAT", "DMIC"; - mux-int-port = <2>; - mux-ext-port = <3>; -}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 37f9b62..6ec19fb 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -231,17 +231,6 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface -config SND_SOC_IMX_WM8962 - tristate "SoC Audio support for i.MX boards with wm8962" - depends on OF && I2C && INPUT - select SND_SOC_WM8962 - select SND_SOC_IMX_PCM_DMA - select SND_SOC_IMX_AUDMUX - select SND_SOC_FSL_SSI - help - Say Y if you want to add support for SoC audio on an i.MX board with - a wm8962 codec. - config SND_SOC_IMX_ES8328 tristate "SoC Audio support for i.MX boards with the ES8328 codec" depends on OF && (I2C || SPI) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index c67bf11..de94fa0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -55,7 +55,6 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o -snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -65,6 +64,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o -obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c deleted file mode 100644 index 206b898..0000000 --- a/sound/soc/fsl/imx-wm8962.c +++ /dev/null @@ -1,312 +0,0 @@ -/* - * Copyright 2013 Freescale Semiconductor, Inc. - * - * Based on imx-sgtl5000.c - * Copyright 2012 Freescale Semiconductor, Inc. - * Copyright 2012 Linaro Ltd. - * - * The code contained herein is licensed under the GNU General Public - * License. You may obtain a copy of the GNU General Public License - * Version 2 or later at the following locations: - * - * http://www.opensource.org/licenses/gpl-license.html - * http://www.gnu.org/copyleft/gpl.html - */ - -#include <linux/module.h> -#include <linux/of_platform.h> -#include <linux/i2c.h> -#include <linux/slab.h> -#include <linux/clk.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> -#include <sound/soc-dapm.h> -#include <linux/pinctrl/consumer.h> - -#include "../codecs/wm8962.h" -#include "imx-audmux.h" - -#define DAI_NAME_SIZE 32 - -struct imx_wm8962_data { - struct snd_soc_dai_link dai; - struct snd_soc_card card; - char codec_dai_name[DAI_NAME_SIZE]; - char platform_name[DAI_NAME_SIZE]; - unsigned int clk_frequency; -}; - -struct imx_priv { - struct platform_device *pdev; - int sample_rate; - snd_pcm_format_t sample_format; -}; - -static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_SPK("Ext Spk", NULL), - SND_SOC_DAPM_MIC("AMIC", NULL), - SND_SOC_DAPM_MIC("DMIC", NULL), -}; - -static int imx_hifi_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_priv *priv = snd_soc_card_get_drvdata(rtd->card); - - priv->sample_rate = params_rate(params); - priv->sample_format = params_format(params); - - return 0; -} - -static const struct snd_soc_ops imx_hifi_ops = { - .hw_params = imx_hifi_hw_params, -}; - -static int imx_wm8962_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - WM8962_FLL_MCLK, data->clk_frequency, - pll_out); - if (ret < 0) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_FLL, pll_out, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level == SND_SOC_BIAS_PREPARE) { - ret = snd_soc_dai_set_sysclk(codec_dai, - WM8962_SYSCLK_MCLK, data->clk_frequency, - SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(dev, - "failed to switch away from FLL: %d\n", - ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, - 0, 0, 0); - if (ret < 0) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - } - break; - - default: - break; - } - - return 0; -} - -static int imx_wm8962_late_probe(struct snd_soc_card *card) -{ - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct imx_priv *priv = snd_soc_card_get_drvdata(card); - struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); - struct device *dev = &priv->pdev->dev; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, - data->clk_frequency, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(dev, "failed to set sysclk in %s\n", __func__); - - return ret; -} - -static int imx_wm8962_probe(struct platform_device *pdev) -{ - struct device_node *np = pdev->dev.of_node; - struct device_node *ssi_np, *codec_np; - struct platform_device *ssi_pdev; - struct i2c_client *codec_dev; - struct imx_wm8962_data *data; - struct imx_priv *priv; - struct clk *codec_clk; - int int_port, ext_port; - int ret; - - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; - - priv->pdev = pdev; - priv->sample_rate = 44100; - priv->sample_format = SNDRV_PCM_FORMAT_S16_LE; - - ret = of_property_read_u32(np, "mux-int-port", &int_port); - if (ret) { - dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); - return ret; - } - ret = of_property_read_u32(np, "mux-ext-port", &ext_port); - if (ret) { - dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); - return ret; - } - - /* - * The port numbering in the hardware manual starts at 1, while - * the audmux API expects it starts at 0. - */ - int_port--; - ext_port--; - ret = imx_audmux_v2_configure_port(int_port, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TCLKDIR, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(&pdev->dev, "audmux internal port setup failed\n"); - return ret; - } - ret = imx_audmux_v2_configure_port(ext_port, - IMX_AUDMUX_V2_PTCR_SYN, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(&pdev->dev, "audmux external port setup failed\n"); - return ret; - } - - ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); - codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); - if (!ssi_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); - ret = -EINVAL; - goto fail; - } - - ssi_pdev = of_find_device_by_node(ssi_np); - if (!ssi_pdev) { - dev_err(&pdev->dev, "failed to find SSI platform device\n"); - ret = -EINVAL; - goto fail; - } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev || !codec_dev->dev.driver) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } - - data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); - if (!data) { - ret = -ENOMEM; - goto fail; - } - - codec_clk = clk_get(&codec_dev->dev, NULL); - if (IS_ERR(codec_clk)) { - ret = PTR_ERR(codec_clk); - dev_err(&codec_dev->dev, "failed to get codec clk: %d\n", ret); - goto fail; - } - - data->clk_frequency = clk_get_rate(codec_clk); - clk_put(codec_clk); - - data->dai.name = "HiFi"; - data->dai.stream_name = "HiFi"; - data->dai.codec_dai_name = "wm8962"; - data->dai.codec_of_node = codec_np; - data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); - data->dai.platform_of_node = ssi_np; - data->dai.ops = &imx_hifi_ops; - data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; - - data->card.dev = &pdev->dev; - ret = snd_soc_of_parse_card_name(&data->card, "model"); - if (ret) - goto fail; - ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); - if (ret) - goto fail; - data->card.num_links = 1; - data->card.owner = THIS_MODULE; - data->card.dai_link = &data->dai; - data->card.dapm_widgets = imx_wm8962_dapm_widgets; - data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); - - data->card.late_probe = imx_wm8962_late_probe; - data->card.set_bias_level = imx_wm8962_set_bias_level; - - platform_set_drvdata(pdev, &data->card); - snd_soc_card_set_drvdata(&data->card, data); - - ret = devm_snd_soc_register_card(&pdev->dev, &data->card); - if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); - goto fail; - } - -fail: - of_node_put(ssi_np); - of_node_put(codec_np); - - return ret; -} - -static const struct of_device_id imx_wm8962_dt_ids[] = { - { .compatible = "fsl,imx-audio-wm8962", }, - { /* sentinel */ } -}; -MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); - -static struct platform_driver imx_wm8962_driver = { - .driver = { - .name = "imx-wm8962", - .pm = &snd_soc_pm_ops, - .of_match_table = imx_wm8962_dt_ids, - }, - .probe = imx_wm8962_probe, -}; -module_platform_driver(imx_wm8962_driver); - -MODULE_AUTHOR("Freescale Semiconductor, Inc."); -MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); -MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:imx-wm8962"); -- 2.7.4

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[alsa-devel] [PATCH v2 0/2] ASoC: add support for ROHM BD28623 codec
by Katsuhiro Suzuki 28 Feb '18

28 Feb '18

This patch adds support for ROHM BD28623MUV class D speaker amplifier codec driver. This driver only refers information of HW specification document that can be derivered at website of ROHM. http://www.rohm.com/web/global/products/-/product/BD28623MUV --- Changes in v2: - Change gpio properties to optional - Change VCCA, VCCP-supply to mandatory - Fix lisence comment style - Use _cansleep() version for manipulating GPIO - Use ASoC component remove() function instead of platform remove() function to power off this IC - Remove update function that was just alias of gpiod_set_value Katsuhiro Suzuki (2): ASoC: add DT bindings documentation for ROHM BD28623 codec ASoC: support ROHM BD28623 codec .../devicetree/bindings/sound/rohm,bd28623.txt | 29 +++ sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/bd28623.c | 242 +++++++++++++++++++++ 4 files changed, 281 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rohm,bd28623.txt create mode 100644 sound/soc/codecs/bd28623.c -- 2.16.1

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