June 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rsrc-card: remove unused ret
by Kuninori Morimoto 17 Jun '15

17 Jun '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> '047000278da3a17f8("ASoC: rsrc-card: cleanup for DPCM")' cleanuped rsrc-card driver, but then, unused ret was left. Below warning happen without this patch ${LINUX}/sound/soc/sh/rcar/rsrc-card.c: In function 'rsrc_card_startup': ${LINUX}/sound/soc/sh/rcar/rsrc-card.c:78:6: warning: unused variable \ 'ret' [-Wunused-variable] Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsrc-card.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sh/rcar/rsrc-card.c b/sound/soc/sh/rcar/rsrc-card.c index 8caca2e..84e9357 100644 --- a/sound/soc/sh/rcar/rsrc-card.c +++ b/sound/soc/sh/rcar/rsrc-card.c @@ -75,8 +75,6 @@ static int rsrc_card_startup(struct snd_pcm_substream *substream) struct rsrc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); struct rsrc_card_dai *dai_props = rsrc_priv_to_props(priv, rtd - rtd->card->rtd); - int ret; - return clk_prepare_enable(dai_props->clk); } -- 1.9.1

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[alsa-devel] [PATCH] ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core
by Vinod Koul 16 Jun '15

16 Jun '15

Since this is common option for HDA driver to specfiy pre-allocated buffer, we should make this option availble to all HDA driver by moving this to HDA core Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/Kconfig | 13 +++++++++++++ sound/pci/hda/Kconfig | 13 ------------- 2 files changed, 13 insertions(+), 13 deletions(-) diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 6dc3914fd28b..3129546398d0 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -14,3 +14,16 @@ config SND_HDA_I915 config SND_HDA_EXT_CORE tristate select SND_HDA_CORE + +config SND_HDA_PREALLOC_SIZE + int "Pre-allocated buffer size for HD-audio driver" + range 0 32768 + default 64 + help + Specifies the default pre-allocated buffer-size in kB for the + HD-audio driver. A larger buffer (e.g. 2048) is preferred + for systems using PulseAudio. The default 64 is chosen just + for compatibility reasons. + + Note that the pre-allocation size can be changed dynamically + via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 98ced971cb43..e94cfd5c69f7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -38,19 +38,6 @@ config SND_HDA_TEGRA if SND_HDA -config SND_HDA_PREALLOC_SIZE - int "Pre-allocated buffer size for HD-audio driver" - range 0 32768 - default 64 - help - Specifies the default pre-allocated buffer-size in kB for the - HD-audio driver. A larger buffer (e.g. 2048) is preferred - for systems using PulseAudio. The default 64 is chosen just - for compatibility reasons. - - Note that the pre-allocation size can be changed dynamically - via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. - config SND_HDA_HWDEP bool "Build hwdep interface for HD-audio driver" select SND_HWDEP -- 1.9.1

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[alsa-devel] [PATCH v5 0/4] ASoC: intel - add skylake PCM driver
by Vinod Koul 16 Jun '15

16 Jun '15

This patch series adds the SKL PCM driver in first two patches and then adds decoupling of the controller for splitting the links in last patch This version removes the bus and match function as that is merged in HDA core now and we have extended HDA bus merged as well Please note this series is dependent on sound next having hda core patches and extended bus patches Jeeja KP (4): ASoC: Intel: add Skylake HDA platform driver ASoC: Intel - add Skylake HDA audio driver ASoC: Intel - add makefile support for SKL driver ASoC: intel - adds support for decoupled mode in skl driver sound/soc/intel/Kconfig | 17 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/skl-pcm.c | 936 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 646 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 74 +++ 6 files changed, 1677 insertions(+) create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/skl-pcm.c create mode 100644 sound/soc/intel/skylake/skl.c create mode 100644 sound/soc/intel/skylake/skl.h -- 1.9.1

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[alsa-devel] ASoC : WM5102 Audio Codec needs ACPI support and a baytrail machine driver
by Christian Hartmann 16 Jun '15

16 Jun '15

Hi list, I have a device "Lenovo Yoga 851 F" which have a WM510205 Audio codec on a Baytrail System. I have sent already one patch (spi-pxa2xx) raise num_chipselect by one, which fixes to bind the arizona WM5102 codec as a spi slave attachted on a pxa2xx spi master device. I have tried several times and added the ACPI id "WM510205" to the arizona-spi.c driver. The ACPI ID is extracted from the DSDT.dsl of this machine. But unfortunately it fails and I got only a NULL pointer deference error in arizona_spi_probe(). I want to request ACPI support for this device and I cannot fix currently the NULL pointer. I do not know yet, if I have to add this ACPI id to the arizona-spi driver (sure, thats what I am trying and failing) or to the NEW baytrail machine driver. In this case it should be the bytcr_wm5102.c and byt-wm5102.c, which does not exists yet. I have a branched tovalds/master and added (my first attempt) the above mentioned machine driver on a local branch. The result is only a dsp boot timeout error. Any help, ideas, hints, patches to try would be appreciate sincerely yours Christian Hartmann a snap of the dsdt Device (LPEA) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "80860F28" /* Intel SST Audio DSP */) // _HID: Hardware ID Name (_CID, "80860F28" /* Intel SST Audio DSP */) // _CID: Compatible ID Name (_DDN, "Intel(R) Low Power Audio Controller - 80860F28") // _DDN: DOS Device Name Name (_SUB, "17AA7004") // _SUB: Subsystem ID Name (_UID, One) // _UID: Unique ID Name (_DEP, Package (0x01) // _DEP: Dependencies { ^SPI1.AUDI }) ... ... Device (AUDI) { Name (_HID, "WM510205") // _HID: Hardware ID Name (_CID, "WM510205") // _CID: Compatible ID Name (_DDN, "Wolfson Microelectronics Audio WM5102") // _DDN: DOS Device Name Method (_CRS, 0, NotSerialized) // _CRS: Current Resource Settings { Name (SBUF, ResourceTemplate () { SpiSerialBus (0x0001, PolarityLow, FourWireMode, 0x08, ControllerInitiated, 0x007A1200, ClockPolarityLow, ClockPhaseFirst, "\\_SB.SPI1", 0x00, ResourceConsumer, , ) GpioInt (Edge, ActiveLow, ExclusiveAndWake, PullNone, 0x0000, "\\_SB.GPO2", 0x00, ResourceConsumer, , ) { // Pin list 0x0004 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.I2C7.PMIC", 0x00, ResourceConsumer, , ) { // Pin list 0x0003 } GpioIo (Exclusive, PullDefault, 0x0000, 0x0000, IoRestrictionOutputOnly, "\\_SB.GPO1", 0x00, ResourceConsumer, , ) { // Pin list 0x0017 } })

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[alsa-devel] [PATCH v2 0/3] ASoC: Mediatek: Add support for MT8173 SOC
by Koro Chen 16 Jun '15

16 Jun '15

This adds basic support for the Mediatek AFE (Audio Front End) unit of MT8173. This patch is based on Linux 4.1-rc1 and Sascha's SCPSYS power patch [1]. The AFE unit comprises several memory interfaces that communicate with CPU, a multi input multi output digital audio interconnect, and several external interfaces (e.g. I2S). [1] https://lkml.org/lkml/2015/6/9/172 changes since v1: - change Kconfig to tristate - remove unnecessary !! in the platform driver - let MODULE_DEVICE_TABLE follow the table - remove regulators from machine drivers changes since RFC: - Use DPCM to model AFE AFE memory interfaces such as DL1, VUL are modeled as frontend DAIs, and AFE IOs such as I2S, HDMI are backend DAIs. Which memif and IO to be used can be set in the machine driver. There is no need to indicate this in dts anymore so include/dt-bindings/sound/mtk-afe.h is removed. AFE connections are represented by DAPM widgets and routes, and set connection APIs/tables are no longer required. (mtk-afe-connection.c and mtk-afe-connection.h are removed)

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[alsa-devel] [PATCH 0/21] ASoC: rsnd: prepare for MIXer support
by Kuninori Morimoto 16 Jun '15

16 Jun '15

Hi Mark These are prepare for MIXer support for Renesas sound card. Each Renesas sound mod (= SSI/SRC/DVC) might be called from many paths if it supports MIXer. In such case, mod <-> stream is no longer 1:1 relationship. this patch set don't use rsnd_mod_to_io() as prepare for MIXer support 1) - 2) bugfix patch for PIO mode. These are not critical bug since no platform is using PIO mode. 3) - 10) cleanup for DPCM related code for MIXer support 11) - 21) remove rsnd_mod_to_io() from Renesas sound driver Kuninori Morimoto (21): 1) ASoC: rsnd: don't care under/over run error when PIO 2) ASoC: rsnd: don't call snd_pcm_period_elapsed() under spin lock 3) ASoC: rsrc-card: used "fe.xxx"/"be.xxx" name for dai_link 4) ASoC: rsrc-card: tidyup priv->snd_card setup timing 5) ASoC: rsrc-card: enable multi cpu/codec for DPCM 6) ASoC: rsrc-card: remove unused name 7) ASoC: rsnd-card: tidyup rsrc_priv_to_() 8) ASoC: rsrc-card: move rsrc_card_parse_daifmt() to upper side 9) ASoC: rsrc-card: tidyup return value of clock error 10) ASoC: rsrc-card: cleanup for DPCM 11) ASoC: rsnd: count each mod (SSI/SRC/DVC) 12) ASoC: rsnd: rsnd_mod has rsnd_priv 13) ASoC: rsnd: adds struct rsnd_dai_stream as on each fuction as parameter 14) ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA 15) ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr() 16) ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx() 17) ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx() 18) ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx() 19) ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol 20) ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working() 21) ASoC: rsnd: remove io from rsnd_mod sound/soc/sh/rcar/core.c | 111 ++++++++++++++++++++++++--------- sound/soc/sh/rcar/dma.c | 113 ++++++++++++++++++++-------------- sound/soc/sh/rcar/dvc.c | 30 +++++---- sound/soc/sh/rcar/rsnd.h | 112 +++++++++++++++++++++------------ sound/soc/sh/rcar/rsrc-card.c | 438 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------------------------------------------------------------------- sound/soc/sh/rcar/src.c | 121 ++++++++++++++++++++---------------- sound/soc/sh/rcar/ssi.c | 96 +++++++++++++++++------------ 7 files changed, 558 insertions(+), 463 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 1/4] ASoC: wm8960: codec mclk should be enabled early to avoid jack detect error
by Zidan Wang 16 Jun '15

16 Jun '15

It will playback from speaker in the first 2 seconds, then switch to headphone. Steps to reproduce this issue: 1. plug out headphone and playback a wav. 2. stop playback and wait for at least 5 seconds, then plug in headphone and playback a wav. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 73 +++++++++++++++++++++-------------------------- 1 file changed, 33 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 761418f..729205f 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -491,6 +491,34 @@ static int wm8960_add_widgets(struct snd_soc_codec *codec) return 0; } +int wm8960_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + int ret; + + if (!IS_ERR(wm8960->mclk)) { + ret = clk_prepare_enable(wm8960->mclk); + if (ret) { + dev_err(codec->dev, + "Failed to enable MCLK: %d\n", ret); + return ret; + } + } + return 0; +} + +void wm8960_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *codec_dai) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); + + if (!IS_ERR(wm8960->mclk)) + clk_disable_unprepare(wm8960->mclk); +} + static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { @@ -702,38 +730,14 @@ static int wm8960_set_bias_level_out3(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int ret; switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: - switch (snd_soc_codec_get_bias_level(codec)) { - case SND_SOC_BIAS_STANDBY: - if (!IS_ERR(wm8960->mclk)) { - ret = clk_prepare_enable(wm8960->mclk); - if (ret) { - dev_err(codec->dev, - "Failed to enable MCLK: %d\n", - ret); - return ret; - } - } - - /* Set VMID to 2x50k */ - snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); - break; - - case SND_SOC_BIAS_ON: - if (!IS_ERR(wm8960->mclk)) - clk_disable_unprepare(wm8960->mclk); - break; - - default: - break; - } - + /* Set VMID to 2x50k */ + snd_soc_update_bits(codec, WM8960_POWER1, 0x180, 0x80); break; case SND_SOC_BIAS_STANDBY: @@ -780,7 +784,7 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm8960_priv *wm8960 = snd_soc_codec_get_drvdata(codec); - int reg, ret; + int reg; switch (level) { case SND_SOC_BIAS_ON: @@ -821,22 +825,9 @@ static int wm8960_set_bias_level_capless(struct snd_soc_codec *codec, WM8960_VREF, WM8960_VREF); msleep(100); - - if (!IS_ERR(wm8960->mclk)) { - ret = clk_prepare_enable(wm8960->mclk); - if (ret) { - dev_err(codec->dev, - "Failed to enable MCLK: %d\n", - ret); - return ret; - } - } break; case SND_SOC_BIAS_ON: - if (!IS_ERR(wm8960->mclk)) - clk_disable_unprepare(wm8960->mclk); - /* Enable anti-pop mode */ snd_soc_update_bits(codec, WM8960_APOP1, WM8960_POBCTRL | WM8960_SOFT_ST | @@ -1059,6 +1050,8 @@ static int wm8960_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8960_dai_ops = { + .startup = wm8960_startup, + .shutdown = wm8960_shutdown, .hw_params = wm8960_hw_params, .digital_mute = wm8960_mute, .set_fmt = wm8960_set_dai_fmt, -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: soc-compress: add a config item for soc-compress
by Jie Yang 16 Jun '15

16 Jun '15

We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, please add 'select SND_SOC_COMPRESS' to driver Kconfig when it is needed. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc.h | 7 +++++++ sound/soc/Kconfig | 6 +++++- sound/soc/Makefile | 3 ++- 3 files changed, 14 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf..03c53a9 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -440,7 +440,14 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +#ifdef CONFIG_SND_SOC_COMPRESS int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#else +static inline int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +{ + return -EPERM; +} +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index e2828e1..de743d9 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,6 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +29,11 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + tristate + select SND_COMPRESS_OFFLOAD + default n + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index a0e1ee6..184c1e6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,7 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o snd-soc-core-objs += soc-topology.o +snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o -- 1.9.1

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[alsa-devel] [PATCH 2/4] ASoC: wm8960: support configure headphone jack detect pin and polarity from device tree
by Zidan Wang 16 Jun '15

16 Jun '15

The ADCLRC/GPIO1, LINPUT3/JD2 and RINPUT3/JD3 pins can be selected as headphone jack detect inputs to automatically disable the speaker output and enable the headphone. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- Documentation/devicetree/bindings/sound/wm8960.txt | 11 +++++++++++ include/sound/wm8960.h | 1 + sound/soc/codecs/wm8960.c | 22 ++++++++++++++++++++++ 3 files changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/wm8960.txt b/Documentation/devicetree/bindings/sound/wm8960.txt index 2deb8a3..4f1aa8b 100644 --- a/Documentation/devicetree/bindings/sound/wm8960.txt +++ b/Documentation/devicetree/bindings/sound/wm8960.txt @@ -21,6 +21,16 @@ Optional properties: enabled and disabled together with HP_L and HP_R pins in response to jack detect events. + - wlf,hp-det: ADCLRC/GPIO1, LINPUT3/JD2 and RINPUT3/JD3 pins can be selected as + headphone jack detect inputs to automatically disable the speaker output + and enable the headphone output. + hp-det = <hp-det-pin hp-det-polarity>; + hp-det-pin = 1: ADCLRC/GPIO1 used as detect pin + hp-det-pin = 2: LINPUT3/JD2 used as detect pin + hp-det-pin = 3: RINPUT3/JD3 used as detect pin + hp-det-polarity = 0: hp detect high for headphone + hp-det-polarity = 1: hp detect high for speaker + Example: codec: wm8960@1a { @@ -28,4 +38,5 @@ codec: wm8960@1a { reg = <0x1a>; wlf,shared-lrclk; + wlf,hp-det = <2 0> }; diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h index e8ce8ee..a787376 100644 --- a/include/sound/wm8960.h +++ b/include/sound/wm8960.h @@ -19,6 +19,7 @@ struct wm8960_data { bool capless; /* Headphone outputs configured in capless mode */ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */ + u32 hp_det[2]; /* headphone detect pin and polarity */ }; #endif diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 729205f..c1be15c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1123,6 +1123,11 @@ static void wm8960_set_pdata_from_of(struct i2c_client *i2c, if (of_property_read_bool(np, "wlf,shared-lrclk")) pdata->shared_lrclk = true; + + of_property_read_u32_array(np, "wlf,hp-det", pdata->hp_det, 2); + + if (pdata->hp_det[0] > 3) + pdata->hp_det[0] = 0; } static int wm8960_i2c_probe(struct i2c_client *i2c, @@ -1168,6 +1173,23 @@ static int wm8960_i2c_probe(struct i2c_client *i2c, } } + /* + * ADCLRC/GPIO1 pin configured as GPIO1 to select jack + * detect input function + */ + if (wm8960->pdata.hp_det[0] == 1) + regmap_update_bits(wm8960->regmap, WM8960_IFACE2, 1<<6, 1<<6); + + /* Enable headphone jack detect */ + if (wm8960->pdata.hp_det[0] != 0) { + regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 1<<6, 1<<6); + regmap_update_bits(wm8960->regmap, WM8960_ADDCTL2, 1<<5, + wm8960->pdata.hp_det[1]<<5); + regmap_update_bits(wm8960->regmap, WM8960_ADDCTL4, 3<<2, + wm8960->pdata.hp_det[0]<<2); + regmap_update_bits(wm8960->regmap, WM8960_ADDCTL1, 3, 3); + } + /* Latch the update bits */ regmap_update_bits(wm8960->regmap, WM8960_LINVOL, 0x100, 0x100); regmap_update_bits(wm8960->regmap, WM8960_RINVOL, 0x100, 0x100); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: DAPM: fix build errors for missing snd_soc_dapm_new_control symbol
by Liam Girdwood 16 Jun '15

16 Jun '15

Fix the following error:- All error/warnings (new ones prefixed by >>): > > sound/built-in.o: In function `soc_tplg_dapm_widget_create': > >> :(.text+0x25a90): undefined reference to `snd_soc_dapm_new_control' Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 92d57a9..1b4a6eb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -53,7 +53,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink)); -static struct snd_soc_dapm_widget * +struct snd_soc_dapm_widget * snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget); @@ -3270,7 +3270,7 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); -static struct snd_soc_dapm_widget * +struct snd_soc_dapm_widget * snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_widget *widget) { -- 2.1.4

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