From e8369d65693bd4e21e043c0e66eff1056ed1e7a3 Mon Sep 17 00:00:00 2001 From: Masanari Iida Date: Fri, 8 Apr 2016 12:45:25 +0900 Subject: ALSA: Fix a typo in timestamping.txt This patch fix a spelling typo found in Documentation/sound/alsa/timestamping.txt Signed-off-by: Masanari Iida Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/timestamping.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/timestamping.txt b/Documentation/sound/alsa/timestamping.txt index 0b191a23f534..1b6473f393a8 100644 --- a/Documentation/sound/alsa/timestamping.txt +++ b/Documentation/sound/alsa/timestamping.txt @@ -129,7 +129,7 @@ will be required to issue multiple queries and perform an interpolation of the results In some hardware-specific configuration, the system timestamp is -latched by a low-level audio subsytem, and the information provided +latched by a low-level audio subsystem, and the information provided back to the driver. Due to potential delays in the communication with the hardware, there is a risk of misalignment with the avail and delay information. To make sure applications are not confused, a -- cgit v1.2.3 From fa44b7ec9bc4115513e59f31da1167166bd6346a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 12 Apr 2016 15:23:05 +0200 Subject: ALSA: hda - Update documentation Update the URLs for alsa-info.sh and hda-emu. Also drop the alsa-driver snapshot URL since it's been discontinued recently. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio.txt | 26 ++++++++------------------ 1 file changed, 8 insertions(+), 18 deletions(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index e7193aac669c..d4510ebf2e8c 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -655,17 +655,6 @@ development branches in general while the development for the current and next kernels are found in for-linus and for-next branches, respectively. -If you are using the latest Linus tree, it'd be better to pull the -above GIT tree onto it. If you are using the older kernels, an easy -way to try the latest ALSA code is to build from the snapshot -tarball. There are daily tarballs and the latest snapshot tarball. -All can be built just like normal alsa-driver release packages, that -is, installed via the usual spells: configure, make and make -install(-modules). See INSTALL in the package. The snapshot tarballs -are found at: - -- ftp://ftp.suse.com/pub/people/tiwai/snapshot/ - Sending a Bug Report ~~~~~~~~~~~~~~~~~~~~ @@ -699,7 +688,12 @@ problems. alsa-info ~~~~~~~~~ The script `alsa-info.sh` is a very useful tool to gather the audio -device information. You can fetch the latest version from: +device information. It's included in alsa-utils package. The latest +version can be found on git repository: + +- git://git.alsa-project.org/alsa-utils.git + +The script can be fetched directly from the following URL, too: - http://www.alsa-project.org/alsa-info.sh @@ -836,15 +830,11 @@ can get a proc-file dump at the current state, get a list of control (mixer) elements, set/get the control element value, simulate the PCM operation, the jack plugging simulation, etc. -The package is found in: - -- ftp://ftp.suse.com/pub/people/tiwai/misc/ - -A git repository is available: +The program is found in the git repository below: - git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/hda-emu.git -See README file in the tarball for more details about hda-emu +See README file in the repository for more details about hda-emu program. -- cgit v1.2.3 From 22225835e23f7a768e767967004d2f3751c64be5 Mon Sep 17 00:00:00 2001 From: Petr Kulhavy Date: Mon, 18 Apr 2016 14:32:40 +0200 Subject: ASoC: davinci-mcbsp: add binding for McBSP Add devicetree binding for the TI DA850/OMAP-L138/AM18xx MultiChannel Buffered Serial Port (McBSP) The optional register range "dat" is not implemented at the moment. The current driver supports only DMA into RX/TX registers but no FIFO. Once the FIFO is implemented in the driver the "dat" range will be used. Signed-off-by: Petr Kulhavy Reviewed-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/davinci-mcbsp.txt | 51 ++++++++++++++++++++++ 1 file changed, 51 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.txt (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt new file mode 100644 index 000000000000..55b53e1fd72c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/davinci-mcbsp.txt @@ -0,0 +1,51 @@ +Texas Instruments DaVinci McBSP module +~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + +This binding describes the "Multi-channel Buffered Serial Port" (McBSP) +audio interface found in some TI DaVinci processors like the OMAP-L138 or AM180x. + + +Required properties: +~~~~~~~~~~~~~~~~~~~~ +- compatible : + "ti,da850-mcbsp" : for DA850, AM180x and OPAM-L138 platforms + +- reg : physical base address and length of the controller memory mapped + region(s). +- reg-names : Should contain: + * "mpu" for the main registers (required). + * "dat" for the data FIFO (optional). + +- dmas: three element list of DMA controller phandles, DMA request line and + TC channel ordered triplets. +- dma-names: identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. The dma + identifiers must be "rx" and "tx". + +Optional properties: +~~~~~~~~~~~~~~~~~~~~ +- interrupts : Interrupt numbers for McBSP +- interrupt-names : Known interrupt names are "rx" and "tx" + +- pinctrl-0: Should specify pin control group used for this controller. +- pinctrl-names: Should contain only one value - "default", for more details + please refer to pinctrl-bindings.txt + +Example (AM1808): +~~~~~~~~~~~~~~~~~ + +mcbsp0: mcbsp@1d10000 { + compatible = "ti,da850-mcbsp"; + pinctrl-names = "default"; + pinctrl-0 = <&mcbsp0_pins>; + + reg = <0x00110000 0x1000>, + <0x00310000 0x1000>; + reg-names = "mpu", "dat"; + interrupts = <97 98>; + interrupts-names = "rx", "tx"; + dmas = <&edma0 3 1 + &edma0 2 1>; + dma-names = "tx", "rx"; + status = "okay"; +}; -- cgit v1.2.3 From 8d84c1973b69621bcb89d5d8a69699e8347a3d90 Mon Sep 17 00:00:00 2001 From: Eric Engestrom Date: Mon, 25 Apr 2016 07:37:02 +0100 Subject: ALSA: doc: fix spelling mistakes Signed-off-by: Eric Engestrom Acked-by: Vinod Koul Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/compress_offload.txt | 2 +- Documentation/sound/alsa/soc/dapm.txt | 2 +- Documentation/sound/alsa/soc/overview.txt | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 630c492c3dc2..81476b4d400b 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -149,7 +149,7 @@ Gapless Playback ================ When playing thru an album, the decoders have the ability to skip the encoder delay and padding and directly move from one track content to another. The end -user can perceive this as gapless playback as we dont have silence while +user can perceive this as gapless playback as we don't have silence while switching from one track to another Also, there might be low-intensity noises due to encoding. Perfect gapless is diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 6faab4880006..c45bd79f291e 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -132,7 +132,7 @@ SOC_DAPM_SINGLE("HiFi Playback Switch", WM8731_APANA, 4, 1, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, wm8731_output_mixer_controls, ARRAY_SIZE(wm8731_output_mixer_controls)), -If you dont want the mixer elements prefixed with the name of the mixer widget, +If you don't want the mixer elements prefixed with the name of the mixer widget, you can use SND_SOC_DAPM_MIXER_NAMED_CTL instead. the parameters are the same as for SND_SOC_DAPM_MIXER. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt index ff88f52eec98..f3f28b7ae242 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -63,7 +63,7 @@ multiple re-usable component drivers :- and any audio DSP drivers for that platform. * Machine class driver: The machine driver class acts as the glue that - decribes and binds the other component drivers together to form an ALSA + describes and binds the other component drivers together to form an ALSA "sound card device". It handles any machine specific controls and machine level audio events (e.g. turning on an amp at start of playback). -- cgit v1.2.3 From 1593af62b694b3638edf577e3b763fa1a4ca3d76 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 4 May 2016 19:33:58 -0300 Subject: ASoC: fsl_sai: Introduce a compatible string for MX6UL MX6UL may need to configure the General Purpose Register 1 (GPR1), so it is better to add a new compatible string to differentiate. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 4 ++-- sound/soc/fsl/fsl_sai.c | 4 +++- 2 files changed, 5 insertions(+), 3 deletions(-) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 044e5d76e2dd..777b941d6cbe 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -7,8 +7,8 @@ codec/DSP interfaces. Required properties: - - compatible : Compatible list, contains "fsl,vf610-sai" or - "fsl,imx6sx-sai". + - compatible : Compatible list, contains "fsl,vf610-sai", + "fsl,imx6sx-sai" or "fsl,imx6ul-sai" - reg : Offset and length of the register set for the device. diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df771e3b..d8b673f7c577 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -797,7 +797,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -898,6 +899,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx6ul-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- cgit v1.2.3 From 4d2458507d0b465c62ae80f3e81b8c008ec96b05 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Wed, 4 May 2016 19:33:59 -0300 Subject: ASoC: fsl_sai: Allow setting the SAI MCLK direction On mx6ul the General Purpose Register 1 (GPR1) contains the following bits for configuring the direction of the SAI MCLKs: SAI1_MCLK_DIR, SAI2_MCLK_DIR, SAI3_MCLK_DIR Introduce the "fsl,sai-mclk-direction-output" optional property to allow configuring the SAI_MCLK outputs. Tested on a imx6ul-evk board. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 5 +++++ include/linux/mfd/syscon/imx6q-iomuxc-gpr.h | 6 ++++++ sound/soc/fsl/fsl_sai.c | 20 ++++++++++++++++++++ 3 files changed, 31 insertions(+) (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 777b941d6cbe..740b467adf7d 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -48,6 +48,11 @@ Required properties: receive data by following their own bit clocks and frame sync clocks separately. +Optional properties (for mx6ul): + + - fsl,sai-mclk-direction-output: This is a boolean property. If present, + indicates that SAI will output the SAI MCLK clock. + Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the default synchronous mode (sync Rx with Tx) will be used, which means both diff --git a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h index 238c8db953eb..68353822afce 100644 --- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h +++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h @@ -447,5 +447,11 @@ #define IMX6UL_GPR1_ENET2_CLK_OUTPUT (0x1 << 18) #define IMX6UL_GPR1_ENET_CLK_DIR (0x3 << 17) #define IMX6UL_GPR1_ENET_CLK_OUTPUT (0x3 << 17) +#define IMX6UL_GPR1_SAI1_MCLK_DIR (0x1 << 19) +#define IMX6UL_GPR1_SAI2_MCLK_DIR (0x1 << 20) +#define IMX6UL_GPR1_SAI3_MCLK_DIR (0x1 << 21) +#define IMX6UL_GPR1_SAI_MCLK_MASK (0x7 << 19) +#define MCLK_DIR(x) (x == 1 ? IMX6UL_GPR1_SAI1_MCLK_DIR : x == 2 ? \ + IMX6UL_GPR1_SAI2_MCLK_DIR : IMX6UL_GPR1_SAI3_MCLK_DIR) #endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */ diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d8b673f7c577..2147994ab46f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -21,6 +21,8 @@ #include #include #include +#include +#include #include "fsl_sai.h" #include "imx-pcm.h" @@ -786,10 +788,12 @@ static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; + struct regmap *gpr; struct resource *res; void __iomem *base; char tmp[8]; int irq, ret, i; + int index; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -878,6 +882,22 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_samplebits = 0; } + if (of_find_property(np, "fsl,sai-mclk-direction-output", NULL) && + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) { + gpr = syscon_regmap_lookup_by_compatible("fsl,imx6ul-iomuxc-gpr"); + if (IS_ERR(gpr)) { + dev_err(&pdev->dev, "cannot find iomuxc registers\n"); + return PTR_ERR(gpr); + } + + index = of_alias_get_id(np, "sai"); + if (index < 0) + return index; + + regmap_update_bits(gpr, IOMUXC_GPR1, MCLK_DIR(index), + MCLK_DIR(index)); + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; -- cgit v1.2.3 From 242658ff99ab9d87e704475ef78c3102ead344cf Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 2 May 2016 14:06:28 +0530 Subject: ALSA: compress: fix some typo Noticed two typos in Documentation, so fix them up Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/compress_offload.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'Documentation') diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 81476b4d400b..8ba556a131c3 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -184,7 +184,7 @@ Sequence flow for gapless would be: - Fill data of the first track - Trigger start - User-space finished sending all, -- Indicaite next track data by sending set_next_track +- Indicate next track data by sending set_next_track - Set metadata of the next track - then call partial_drain to flush most of buffer in DSP - Fill data of the next track -- cgit v1.2.3 From 97d3ddd71fbf663a5da52897757333341a8b254f Mon Sep 17 00:00:00 2001 From: Florian Meier Date: Fri, 13 May 2016 09:14:12 +0000 Subject: ASoC: pcm5102a: Add support for PCM5102A codec Some definitions to support the PCM5102A codec by Texas Instruments. Signed-off-by: Florian Meier Changes to original patch by Florian Meier: * rebased (Makefile and Kconfig * fixed checkpath errors (spaces, newlines) * added dt-binding documentation Signed-off-by: Martin Sperl Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/pcm5102a.txt | 13 ++++ sound/soc/codecs/Kconfig | 4 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm5102a.c | 69 ++++++++++++++++++++++ 4 files changed, 88 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm5102a.txt create mode 100644 sound/soc/codecs/pcm5102a.c (limited to 'Documentation') diff --git a/Documentation/devicetree/bindings/sound/pcm5102a.txt b/Documentation/devicetree/bindings/sound/pcm5102a.txt new file mode 100644 index 000000000000..c63ab0b6ee19 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm5102a.txt @@ -0,0 +1,13 @@ +PCM5102a audio CODECs + +These devices does not use I2C or SPI. + +Required properties: + + - compatible : set as "ti,pcm5102a" + +Examples: + + pcm5102a: pcm5102a { + compatible = "ti,pcm5102a"; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..f736953a4fd9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -94,6 +94,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_PCM3168A_I2C if I2C select SND_SOC_PCM3168A_SPI if SPI_MASTER + select SND_SOC_PCM5102A select SND_SOC_PCM512x_I2C if I2C select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT286 if I2C @@ -575,6 +576,9 @@ config SND_SOC_PCM3168A_SPI select SND_SOC_PCM3168A select REGMAP_SPI +config SND_SOC_PCM5102A + tristate + config SND_SOC_PCM512x tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..4532a743b5f8 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -89,6 +89,7 @@ snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm3168a-objs := pcm3168a.o snd-soc-pcm3168a-i2c-objs := pcm3168a-i2c.o snd-soc-pcm3168a-spi-objs := pcm3168a-spi.o +snd-soc-pcm5102a-objs := pcm5102a.o snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o @@ -298,6 +299,7 @@ obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM3168A) += snd-soc-pcm3168a.o obj-$(CONFIG_SND_SOC_PCM3168A_I2C) += snd-soc-pcm3168a-i2c.o obj-$(CONFIG_SND_SOC_PCM3168A_SPI) += snd-soc-pcm3168a-spi.o +obj-$(CONFIG_SND_SOC_PCM5102A) += snd-soc-pcm5102a.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c new file mode 100644 index 000000000000..ed515677409b --- /dev/null +++ b/sound/soc/codecs/pcm5102a.c @@ -0,0 +1,69 @@ +/* + * Driver for the PCM5102A codec + * + * Author: Florian Meier + * Copyright 2013 + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include + +#include + +static struct snd_soc_dai_driver pcm5102a_dai = { + .name = "pcm5102a-hifi", + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_pcm5102a; + +static int pcm5102a_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pcm5102a, + &pcm5102a_dai, 1); +} + +static int pcm5102a_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static const struct of_device_id pcm5102a_of_match[] = { + { .compatible = "ti,pcm5102a", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm5102a_of_match); + +static struct platform_driver pcm5102a_codec_driver = { + .probe = pcm5102a_probe, + .remove = pcm5102a_remove, + .driver = { + .name = "pcm5102a-codec", + .owner = THIS_MODULE, + .of_match_table = pcm5102a_of_match, + }, +}; + +module_platform_driver(pcm5102a_codec_driver); + +MODULE_DESCRIPTION("ASoC PCM5102A codec driver"); +MODULE_AUTHOR("Florian Meier "); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3