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

[alsa-devel] [asoc:topic/gtm601 2/3] sound/soc/codecs/gtm601.c:86:4-9: No need to set .owner here. The core will do it.
by kbuild test robot 22 Jun '15

22 Jun '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/gtm601 head: ac6e7c29ac12e1e285b0740bb671039ef436cf91 commit: 600a2b098a87843d44ab2f54fde9eda421946b38 [2/3] ASoC: Add gtm601 codec driver coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/gtm601.c:86:4-9: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 1

[alsa-devel] [PATCH] ASoC: rsrc-card: Remove unused variable in rsrc_card_startup()
by Geert Uytterhoeven 22 Jun '15

22 Jun '15

sound/soc/sh/rcar/rsrc-card.c: In function 'rsrc_card_startup': sound/soc/sh/rcar/rsrc-card.c:78:6: warning: unused variable 'ret' [-Wunused-variable] int ret; ^ Fixes: 047000278da3a17f ("ASoC: rsrc-card: cleanup for DPCM") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- 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 8caca2e180c39122..84e935711e29e442 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

5 7

[alsa-devel] [PATCH v2 0/2] ALSA: hda: add module flag snd_adsp_enable
by Vinod Koul 22 Jun '15

22 Jun '15

This is second attempt for creating a module option for hda core to select either legacy HDA driver or upcoming ASoC driver. This series still relies on module flag (with feedback from previous series baked in). The eventual plan is to rely on HW advertised capabilities and BIOS configuration to load either driver, but since ASoC driver has still some time to get fully merged and matured on various system, we will rely on module flag till the who ASoC HDA systems is completed and available is all distros Vinod Koul (2): ALSA: hda: add snd_adsp_enable module flag ALSA: hda: use snd_adsp_enable in HDA driver include/sound/hdaudio.h | 11 +++++++++++ sound/hda/hda_bus_type.c | 6 ++++++ sound/pci/hda/hda_controller.h | 1 + sound/pci/hda/hda_intel.c | 10 +++++++++- 4 files changed, 27 insertions(+), 1 deletion(-) -- 1.9.1

2 4

[alsa-devel] Post Audio codec sgtl5000 with ARM/Zynq under Linux
by Wojciech Zebrowski 22 Jun '15

22 Jun '15

Hi All I try solve problem with support SGTL5000 under Linux , ARM / ZYNQ ARCHITECTURE. Please add this post to "alsa" forum. If you need any others information i can send it . My post: I try use freescale audio chip sgtl5000 on my custom board with Zynq chip. I need help with instance driver in device tree and debug first execution. 1.*I use I2C* controler to configure sgtl5000 : / axi_iic_sgtl: i2c@41610000 {// // #address-cells = <1>;// // #size-cells = <0>;// // compatible = "xlnx,xps-iic-2.00.a";// // interrupt-parent = <&gic>;// // interrupts = <0 31 4>;// // reg = <0x41610000 0x10000>;// //// // codec: sgtl5000@0a {// // #sound-dai-cells = <0>;// // compatible = "fsl,sgtl5000";// // reg = <0x0a>;// ////// VDDA-supply = <&reg_3p3v>;// ////// VDDIO-supply = <&reg_3p3v>;// // clocks = <&clkc 19>;// // }; / How can i set comment lines ? 2. sound : / sound { compatible = "fsl,imx51-babbage-sgtl5000", "fsl,imx-audio-sgtl5000"; model = "imx51-babbage-sgtl5000"; // ssi-controller = <&ssi1>; audio-codec = <&codec>; audio-routing = "MIC_IN", "Mic Jack", "Mic Jack", "Mic Bias", "Headphone Jack", "HP_OUT"; mux-int-port = <1>; mux-ext-port = <3>; };/ I want use I2C configuration driver not SPI . How can i set comment line ? 3. I2S driver looks like ok : / axi_i2s_adi_0: axi_i2s_adi@77600000 { compatible = "xlnx,axi-i2s-adi-1.0"; reg = <0x77600000 0x10000>; xlnx,bclk-pol = <0x0>; xlnx,dma-type = <0x1>; xlnx,has-rx = <0x1>; xlnx,has-tx = <0x1>; xlnx,lrclk-pol = <0x0>; xlnx,num-ch = <0x1>; xlnx,s-axi-min-size = <0x000001FF>; xlnx,slot-width = <0x18>; };/ PS. Right now i can compile device tree but audio is not detected. What kind of log do you need to check Linux boot problem ?

2 3

[alsa-devel] [PATCH v6 0/4] ASoC: intel - add skylake PCM driver
by Vinod Koul 22 Jun '15

22 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 stream object creation, freeup, link freeup and buffer prealloc flag as they have moved to hdac now. Also irq management is improved upon. Please note this series is dependent on sound next having hda core patches and extended bus patches Jeeja KP (3): ASoC: Intel: add Skylake HDA platform driver ASoC: Intel - add Skylake HDA audio driver ASoC: Intel - add makefile support for SKL driver Vinod Koul (1): ASoC: intel - adds support for decoupled mode in skl driver sound/soc/intel/Kconfig | 7 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/skl-pcm.c | 935 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 543 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 71 +++ 6 files changed, 1560 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

2 10

[alsa-devel] [PATCH] ASoC: fsl: fsl_asrc: Check for clk_prepare_enable() error
by Fabio Estevam 20 Jun '15

20 Jun '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> clk_prepare_enable() may fail, so we should better check its return value and propagate it in the case of error. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_asrc.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index c068494..37dde26 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -931,14 +931,22 @@ static int fsl_asrc_probe(struct platform_device *pdev) static int fsl_asrc_runtime_resume(struct device *dev) { struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); - int i; + int i, ret; - clk_prepare_enable(asrc_priv->mem_clk); - clk_prepare_enable(asrc_priv->ipg_clk); + ret = clk_prepare_enable(asrc_priv->mem_clk); + if (ret) + return ret; + ret = clk_prepare_enable(asrc_priv->ipg_clk); + if (ret) + goto disable_mem_clk; for (i = 0; i < ASRC_CLK_MAX_NUM; i++) clk_prepare_enable(asrc_priv->asrck_clk[i]); return 0; + +disable_mem_clk: + clk_disable_unprepare(asrc_priv->mem_clk); + return ret; } static int fsl_asrc_runtime_suspend(struct device *dev) -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda-beep: Update authors dead email address
by Matt Ranostay 20 Jun '15

20 Jun '15

Signed-off-by: Matt Ranostay <mranostay(a)gmail.com> --- sound/pci/hda/hda_beep.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 46524ff..1052ad3 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -1,7 +1,7 @@ /* * Digital Beep Input Interface for HD-audio codec * - * Author: Matthew Ranostay <mranostay(a)embeddedalley.com> + * Author: Matt Ranostay <mranostay(a)gmail.com> * Copyright (c) 2008 Embedded Alley Solutions Inc * * This driver is free software; you can redistribute it and/or modify -- 1.9.1

2 1

Re: [alsa-devel] No sound from Mac Pro headphone port
by Raymond Yau 20 Jun '15

20 Jun '15

> > You will need to provide more specific instructions on what you want. I do not read code. > https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> >> Bingo: the pin-outs are very incorrect. >> >> This is the (original) tech specs of the machine. >> >> https://support.apple.com/kb/SP506?viewlocale=en_US&locale=en_US >> >> Note: >> >> 1 x internal speaker >> 1 x FRONT headphone port >> >> >> Refer to user manual , >> Headphone Jack >> Connect headphones, including Apple Earphone with remote and mic through >> the headphone mini jack >> >> Is node 0x18 headset mic ? >> need specific sequence number and parser_flags >> >> https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… >> >> +#define HDA_PINCFG_HEADSET_MIC (1 << 2) /* Try to find headset mic; >> mark seq number as 0xc to trigger */ >> >> 1 x REAR line out port >> >> 1 x REAR line in port >> 1 x REAR optical TOSLINK out port >> 1 x REAR optical TOSLINK in port >> >> The machine does not have a built-in mic port. >> >> >> you need to use hdajacksensetest to verify hp, line out , line in jacks >> are node 0x19, 0x14 and 0x1a by plug and unplug those jacks >> >> clear bit 8 of pin default of Line Out Jack if it return correct jack state >> >> >> >> Linux 4. Mac Pro 2009. ACL889A codec. >> >> No sound from headphone port. Seems to detect plug / unplug OK. >> >> Works OK when machine booted to OS X or Windows 7. >> >> Have found several bug repors on similar issues with Mac >> laptops but >> no fixes. >> >> Have written e-mail with David Henningsson on problem but >> unable to >> find a solution. >> >> ALSA-info output: >> >> http://www.alsa-project.org/db/?f=1dc7146eb34af5ad307d029868f775fac5e2764e >> >> >> Do your macpro spec support 5.1 ? >> >> Simple mixer control 'Channel Mode',0 >> Capabilities: enum >> Items: '2ch' '4ch' '6ch' >> Item0: '2ch' Channel mode is created when there are three or more jacks at same location Use hdajackretask to change location of HP and Mic from Ext Rear to Ext Front Advanced override and set as boot default http://git.alsa-project.org/?p=alsa-tools.git;a=blob;f=hdajackretask/README… http://voices.canonical.com/david.henningsson/2011/11/29/turn-your-mic-jack… http://git.alsa-project.org/?p=alsa-tools.git;a=tree;f=hdajacksensetest;hb=… Use hdajacksensetest -a to verify line out support jack detect clear MISC = NO_PRESENCE bit 8 of pin default of Line out by hdajackretask Pin Default 0x01010050: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Unknown DefAssociation = 0x5, Sequence = 0x0 Most likely need vendor coefficient to support mono mic of apple earphones >> >> Pin default seem incorrect >> >> 1) Headphone at ext Rear >> 2) Line Out does not support jack detection >> 3) Mic jack ???? >> >> /sys/class/sound/hwC0D0/init_pin_configs: >> 0x14 0x01010150 >> 0x15 0x400000f0 >> 0x16 0x400000f0 >> 0x17 0x400000f0 >> 0x18 0x01a19070 >> 0x19 0x01214040 >> 0x1a 0x01813010 >> 0x1b 0x90100130 >> 0x1c 0x400000f0 >> 0x1d 0x400000f0 >> 0x1e 0x0145e160 >> 0x1f 0x01c5e120 >> >> /sys/class/sound/hwC0D0/driver_pin_configs: >> >> >> >> [ 4.057250] sound hdaudioC0D0: ALC889A: SKU not ready 0x400000f0 >> [ 4.057633] sound hdaudioC0D0: autoconfig for ALC889A: >> line_outs=1 >> (0x14/0x0/0x0/0x0/0x0) type:line >> [ 4.057636] sound hdaudioC0D0: speaker_outs=1 >> (0x1b/0x0/0x0/0x0/0x0) >> [ 4.057638] sound hdaudioC0D0: hp_outs=1 >> (0x19/0x0/0x0/0x0/0x0) >> [ 4.057639] sound hdaudioC0D0: mono: mono_out=0x0 >> [ 4.057641] sound hdaudioC0D0: dig-out=0x1e/0x0 >> [ 4.057642] sound hdaudioC0D0: inputs: >> [ 4.057645] sound hdaudioC0D0: Mic=0x18 >> [ 4.057646] sound hdaudioC0D0: Line=0x1a >> [ 4.057648] sound hdaudioC0D0: dig-in=0x1f >> >>

2 8

[alsa-devel] Scarlett 18i8 hardware out of sync with AlsaMixer
by Adam Goode 19 Jun '15

19 Jun '15

Linux 4.0.2-300.fc22.x86_64 Hi, I have the Scarlett 18i8 USB device that is supported by the new scarlett_mixer.c code. I am happy to say that the mixer code works in most cases. But under some conditions, I cannot get any sound out of the device until I go and toggle the "Matrix 01 Input" and "Matrix 02 Input" enums up and then down (from PCM 1 and PCM 2). This appears to me to be some kind of cache invalidation bug, where the device is out of sync with kernel mixer state. I will take a look at the code myself at some point, but not for a while. But I am happy to try patches if anyone comes up with anything in the mean time. Thanks, Adam

4 9

[alsa-devel] [PATCH] ASoC: wm_adsp: Move DSP Rate controls into the codec
by Richard Fitzgerald 19 Jun '15

19 Jun '15

The rate controls are codec-specific, it's not possible to generically say what the range or the meaning of each control is (or even if they exist at all) - that depends on the particular codec. This is currently being handled for Arizona codecs by putting an Arizona-specific table of controls inside the wm_adsp driver. This creates a dependency between wm_adsp and arizona.c, and is an awkward solution if the ADSP is used in another family of codecs Fix this by moving the Arizona-specific rate controls into the Arizona codec drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 27 ++++++++++++++++++++++ sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 5 ++++ sound/soc/codecs/wm5110.c | 6 +++++ sound/soc/codecs/wm_adsp.c | 52 +++++-------------------------------------- sound/soc/codecs/wm_adsp.h | 2 +- 7 files changed, 48 insertions(+), 48 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 88f6df2..802e05e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -2286,6 +2286,33 @@ int arizona_set_output_mode(struct snd_soc_codec *codec, int output, bool diff) } EXPORT_SYMBOL_GPL(arizona_set_output_mode); +static const struct soc_enum arizona_adsp2_rate_enum[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP2_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP4_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; + +const struct snd_kcontrol_new arizona_adsp2_rate_controls[] = { + SOC_ENUM("DSP1 Rate", arizona_adsp2_rate_enum[0]), + SOC_ENUM("DSP2 Rate", arizona_adsp2_rate_enum[1]), + SOC_ENUM("DSP3 Rate", arizona_adsp2_rate_enum[2]), + SOC_ENUM("DSP4 Rate", arizona_adsp2_rate_enum[3]), +}; +EXPORT_SYMBOL_GPL(arizona_adsp2_rate_controls); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 8824f5b..9881b6f 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -217,6 +217,8 @@ extern const struct soc_enum arizona_ng_hold; extern const struct soc_enum arizona_in_hpf_cut_enum; extern const struct soc_enum arizona_in_dmic_osr[]; +extern const struct snd_kcontrol_new arizona_adsp2_rate_controls[]; + extern int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 5a9da28..c830832 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1555,7 +1555,7 @@ static int wm2200_probe(struct snd_soc_codec *codec) wm2200->codec = codec; - ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); + ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index c5ec519..d097f09e5 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1880,6 +1880,11 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret) return ret; + ret = snd_soc_add_codec_controls(codec, + arizona_adsp2_rate_controls, 1); + if (ret) + return ret; + arizona_init_spk(codec); arizona_init_gpio(codec); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 5f032a3..709fcc6 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1614,6 +1614,12 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) return ret; } + ret = snd_soc_add_codec_controls(codec, + arizona_adsp2_rate_controls, + WM5110_NUM_ADSP); + if (ret) + return ret; + snd_soc_dapm_disable_pin(dapm, "HAPTICS"); return 0; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f9f90b0..0bb415a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -458,57 +458,17 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 3, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; -const struct snd_kcontrol_new wm_adsp1_fw_controls[] = { +const struct snd_kcontrol_new wm_adsp_fw_controls[] = { SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], wm_adsp_fw_get, wm_adsp_fw_put), SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], wm_adsp_fw_get, wm_adsp_fw_put), SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], + wm_adsp_fw_get, wm_adsp_fw_put), }; -EXPORT_SYMBOL_GPL(wm_adsp1_fw_controls); - -static const struct soc_enum wm_adsp2_rate_enum[] = { - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP2_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), - SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP4_CONTROL_1, - ARIZONA_DSP1_RATE_SHIFT, 0xf, - ARIZONA_RATE_ENUM_SIZE, - arizona_rate_text, arizona_rate_val), -}; - -static const struct snd_kcontrol_new wm_adsp2_fw_controls[4][2] = { - { - SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP1 Rate", wm_adsp2_rate_enum[0]), - }, - { - SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP2 Rate", wm_adsp2_rate_enum[1]), - }, - { - SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP3 Rate", wm_adsp2_rate_enum[2]), - }, - { - SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], - wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP4 Rate", wm_adsp2_rate_enum[3]), - }, -}; +EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) @@ -2125,8 +2085,8 @@ int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec) wm_adsp2_init_debugfs(dsp, codec); return snd_soc_add_codec_controls(codec, - wm_adsp2_fw_controls[dsp->num - 1], - ARRAY_SIZE(wm_adsp2_fw_controls[0])); + &wm_adsp_fw_controls[dsp->num - 1], + 1); } EXPORT_SYMBOL_GPL(wm_adsp2_codec_probe); diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 5042cbd..579a635 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -83,7 +83,7 @@ struct wm_adsp { #define WM_ADSP2(wname, num) \ WM_ADSP2_E(wname, num, wm_adsp2_early_event) -extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; +extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); -- 1.7.2.5

2 1