November 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] mic bias for tlv320aic3x?
by Peter Meerwald 28 Nov '12

28 Nov '12

Hello, I'd like to enable 2V mic bias on the tlv320aic3x (which offers 2V, 2.5V, AVDD, off) -- I'd have expected some control in alsamixer to so, but there is none what is the proper way to implement mic bias selection? thanks, regards, p. -- Peter Meerwald +43-664-2444418 (mobile)

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[alsa-devel] [PATCH 1/1] ASoC: wm2200: Remove empty labels
by Sachin Kamat 28 Nov '12

28 Nov '12

Removed unnecessary labels probably left out of some previous cleanup. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/wm2200.c | 9 +++------ 1 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 1730df8..ff45b02 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -2205,7 +2205,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm2200->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - goto err; + return ret; } for (i = 0; i < 2; i++) { @@ -2238,7 +2238,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to request core supplies: %d\n", ret); - goto err_regmap; + return ret; } ret = regulator_bulk_enable(ARRAY_SIZE(wm2200->core_supplies), @@ -2246,7 +2246,7 @@ static __devinit int wm2200_i2c_probe(struct i2c_client *i2c, if (ret != 0) { dev_err(&i2c->dev, "Failed to enable core supplies: %d\n", ret); - goto err_core; + return ret; } if (wm2200->pdata.ldo_ena) { @@ -2382,9 +2382,6 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm2200->core_supplies), wm2200->core_supplies); -err_core: -err_regmap: -err: return ret; } -- 1.7.4.1

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Re: [alsa-devel] [GIT PULL] ASoC: Samsung: Updates for v3.8
by Padma Venkat 28 Nov '12

28 Nov '12

Hi Mark, On Wed, Nov 28, 2012 at 10:55 AM, Sangbeom Kim <sbkim73(a)samsung.com> wrote: >> > There's some problems with this binding. The main one is the gpios >> > property the format of which isn't specified at all. > >> All of above gpio property is i2s. >> That is, >> + gpios = <&gpz 0 2 0 0>, -> SCLK >> + <&gpz 1 2 0 0>, -> CDCLK >> + <&gpz 2 2 0 0>, -> LRCK >> + <&gpz 3 2 0 0>, -> SDI >> + <&gpz 4 2 0 0>, -> SDO[0] >> + <&gpz 5 2 0 0>, -> SDO[1] >> + <&gpz 6 2 0 0>; -> SDO[2] > >> Do you want like a below one? >> +sclk-gpios = <&gpz 0 2 0 0>, >> +cdclk-gpios = <&gpz 1 2 0 0>, ... > > That would be nice but what I was really looking for was some indiciation in the binding document as > to what all this stuff means - it should really say what gpz is (though I can figure that out) and > it definitely needs to say what the four numbers are. > >> > The requirement for an alias is also very odd, where does that come from? > >> I don't know that Which one is odd. Please let me know. > > Having them at all is odd - it's not something other DT bindings have needed and there was nothing > saying what it was for. What is the exact requirement here? Should I remove the alias ids or I2S1 and I2S2 nodes as they are in disabled state? This is done same way as SPI and I2C. Please explain me. > >> > Some of the code also looks very peculiar, like the fact that it's >> > generating a clock name i2s_opclk%d rather than hard coding the >> > clock, the physical clock would normally be resolved based on the >> > struct device. > >> This is to handle all of Samsung SOCs i2c clock mux. >> Please look at below clk_lookup table > >> In case of 6410, clk_lookup >> + CLKDEV_INIT("samsung-i2s.0", "i2s_opclk0", &clk_i2s0), >> + CLKDEV_INIT("samsung-i2s.0", "i2s_opclk1", &clk_audio_bus0.clk), >> + CLKDEV_INIT("samsung-i2s.1", "i2s_opclk0", &clk_i2s1), >> + CLKDEV_INIT("samsung-i2s.1", "i2s_opclk1", &clk_audio_bus1.clk), >> +#ifdef CONFIG_CPU_S3C6410 >> + CLKDEV_INIT("samsung-i2s.2", "i2s_opclk0", &clk_i2s2), >> + CLKDEV_INIT("samsung-i2s.2", "i2s_opclk1", &clk_audio_bus2.clk), > >> In case of exynos5, clk_lookup >> + CLKDEV_INIT("samsung-i2s.0", "i2s_opclk0", &exynos5_clk_sclk_i2s.clk), >> + CLKDEV_INIT("samsung-i2s.0", "i2s_opclk1", >> +&exynos5_clk_i2s_bus.clk), > >> We try to handle clock source of i2s by only i2s_opclk0 and i2s_opclk1. >> Each SOCs have different clock source. >> Is this wrong approach? > > That makes sense but why is the driver not just requesting by name rather than having code to > generate the names, and why is this mixed in with the patch adding DT support? > Thanks Padma

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[alsa-devel] [PATCH 1/1] ALSA: CA0132 - Handle endianness of SCP buffer data
by Ian Minett 28 Nov '12

28 Nov '12

From: Ian Minett <ian_minett(a)creativelabs.com> Handle endianness of the SCP buffer being sent to or from the DSP chip on big-endian host architectures. The SCP buffer is prepared in dspio_scp(), and bytes are swapped before being transferred by the codec in dspio_send(). Based on branch origin/topic/hda-ca0132-dsp. We would appreciate any feedback you may have. Thanks very much, - Ian Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2fd3121..8386a11 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -28,6 +28,7 @@ #include <linux/mutex.h> #include <linux/module.h> #include <linux/firmware.h> +#include <asm/unaligned.h> #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" @@ -701,11 +702,16 @@ static int dspio_send(struct hda_codec *codec, unsigned int reg, unsigned int res; int retry = 50; + /* Handling for Big Endian machine */ + unsigned int val; + put_unaligned_le32(data, &val); + /* send bits of data specified by reg to dsp */ do { - res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, data); - if ((res >= 0) && (res != VENDOR_STATUS_DSPIO_BUSY)) - return res; + res = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, reg, val); + put_unaligned_le32(res, &val); + if ((val >= 0) && (val != VENDOR_STATUS_DSPIO_BUSY)) + return val; } while (--retry); return -EIO; @@ -719,11 +725,15 @@ static void dspio_write_wait(struct hda_codec *codec) int status; unsigned long timeout = jiffies + msecs_to_jiffies(1000); + /* Handling for Big Endian machine */ + unsigned int val; + do { status = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, VENDOR_DSPIO_STATUS, 0); - if ((status == VENDOR_STATUS_DSPIO_OK) || - (status == VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY)) + put_unaligned_le32(status, &val); + if ((val == VENDOR_STATUS_DSPIO_OK) || + (val == VENDOR_STATUS_DSPIO_SCP_RESPONSE_QUEUE_EMPTY)) break; msleep(1); } while (time_before(jiffies, timeout)); @@ -736,6 +746,8 @@ static int dspio_write(struct hda_codec *codec, unsigned int scp_data) { struct ca0132_spec *spec = codec->spec; int status; + /* Handling for Big Endian machine */ + unsigned int val = 0; dspio_write_wait(codec); @@ -751,8 +763,10 @@ static int dspio_write(struct hda_codec *codec, unsigned int scp_data) goto error; /* OK, now check if the write itself has executed*/ - status = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, + val = snd_hda_codec_read(codec, WIDGET_DSP_CTRL, 0, VENDOR_DSPIO_STATUS, 0); + put_unaligned_le32(val, &status); + error: mutex_unlock(&spec->chipio_mutex); -- 1.7.4.1

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[alsa-devel] Missing exactly 3 of 8 audio packets?
by Daniel Griscom 27 Nov '12

27 Nov '12

On to the next problem we're having with our USB audio card. Recap: simple USB Audio Class embedded stereo audio card, running at 44.1kHz or 48kHz, using the 3.6.6 kernel and its associated ALSA modules to drive it, running on Jetway ND9D-2700, with an Atom D2700 and NM10 chipset. Output of "sudo lsusb -v" for card is appended below. Problem: we're experiencing strange hiccups in the delivery of audio from the motherboard to the card. At 48kHz, the motherboard should send a 6-frame packet to the card once per high-speed microframe (every 125us). This usually works, but sometimes there are two closely related failures: 1) Once in a while (every few seconds or minutes) a single packet will be missed 2) Once in a longer while (every few minutes or hours), the system will go into a continuing error condition where three of every eight packets are missed. In both cases, "missed" means that ALSA consumes the output audio from my motherboard application, but never sends a packet to the audio card. No USB errors, no logged messages; it just disappears. The second case has one very interesting symptom. The pattern of missed frames is amazingly rigid, and repeats the following pattern endlessly (each character is a 125us slot for a packet, "F" is a transmitted packet, "-" is a missing packet): FFFF--F- So, with eight microframes per millisecond where packets can be sent, the fifth, sixth and eighth packets are missed, and this repeats for at least a number of seconds. Very, very strange. Using a Beagle 1200 I can see that there are no USB errors: the packet is just not sent. Audio continues unabated in the other direction. Timing always looks good, no other traffic on that USB port. We've been chewing on this issue for a while. To me, it sounds like this could only happen in the ALSA code where the stream of audio is broken into individual USB packets. Unfortunately, I'm having a heck of a time navigating the ALSA source code, and frankly have gotten no traction at all. Any thoughts on what this might be, or on what further tests we might do? Thanks, Dan P.S. Here's the output of "sudo lsusb -v" for the device. Bus 001 Device 005: ID 03eb:2311 Atmel Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x03eb Atmel Corp. idProduct 0x2311 bcdDevice 10.00 iManufacturer 1 Suitable Systems iProduct 2 USB 24-bit Stereo AudioCard iSerial 3 1.0.0.0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 266 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 107 bInCollection 2 baInterfaceNr( 0) 2 baInterfaceNr( 1) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 1 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 17 AES/EBU AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 14 wTerminalType 0x0602 Digital Audio Interface bAssocTerminal 0 bNrChannels 1 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 18 SPDIF AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 1 bControlSize 1 bmaControls( 0) 0x03 Mute Volume bmaControls( 1) 0x00 iFeature 10 Output AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 2 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 1 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 7 Analog AudioControl Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 8 bSourceID 4 bControlSize 1 bmaControls( 0) 0x00 bmaControls( 1) 0x80 Delay bmaControls( 2) 0x80 Delay bmaControls( 3) 0x80 Delay bmaControls( 4) 0x80 Delay bmaControls( 5) 0x80 Delay bmaControls( 6) 0x80 Delay bmaControls( 7) 0x80 Delay bmaControls( 8) 0x80 Delay iFeature 12 ADC Regs AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 7 bNrInPins 3 baSource( 0) 8 baSource( 1) 1 baSource( 2) 14 iSelector 11 Source Select AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 7 bControlSize 1 bmaControls( 0) 0x03 Mute Volume bmaControls( 1) 0x00 iFeature 9 Input AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 5 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 3 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 1 bRefresh 0 bSynchAddress 2 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 6 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 3 Discrete tSamFreq[ 0] 32000 tSamFreq[ 1] 44100 tSamFreq[ 2] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0100 1x 256 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 1 Keyboard iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 11.01 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 39 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0008 1x 8 bytes bInterval 2 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered -- Daniel T. Griscom griscom(a)suitable.com Suitable Systems http://www.suitable.com/ 1 Centre Street, Suite 204 (781) 665-0053 Wakefield, MA 01880-2400

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Re: [alsa-devel] [BUGFIX PATCH] ASoC: fsl: fix miscompilation of snd-soc-imx-pcm
by Mark Brown 27 Nov '12

27 Nov '12

On Thu, Nov 22, 2012 at 01:31:06PM +0100, Lothar Waßmann wrote: > obj-$(CONFIG_SND_SOC_IMX_PCM) += snd-soc-imx-pcm.o > snd-soc-imx-pcm-y := imx-pcm.o > -snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm-fiq.o > -snd-soc-imx-pcm-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm-dma.o > +ifneq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),) > + snd-soc-imx-pcm-objs += imx-pcm-fiq.o > +endif > +ifneq ($(CONFIG_SND_SOC_IMX_PCM_DMA),) > + snd-soc-imx-pcm-objs += imx-pcm-dma.o > +endif What is the actual bug here? This fix doesn't look obviously right, and if we do need to move to -objs for some reason then the block ought to be in the -objs section of the Makefile.

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[alsa-devel] [PATCH] ASoC: wm5102: Add missing routes for ASRC inputs
by Mark Brown 27 Nov '12

27 Nov '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 63c24a3..5a1eb5b 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1221,6 +1221,11 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "IN3L PGA", NULL, "IN3L" }, { "IN3R PGA", NULL, "IN3R" }, + { "ASRC1L", NULL, "ASRC1L Input" }, + { "ASRC1R", NULL, "ASRC1R Input" }, + { "ASRC2L", NULL, "ASRC2L Input" }, + { "ASRC2R", NULL, "ASRC2R Input" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: pcm: Report error code when we fail to init platform
by Mark Brown 27 Nov '12

27 Nov '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 314102a..5c3ca2a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2100,7 +2100,9 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { - dev_err(platform->dev, "ASoC: pcm constructor failed\n"); + dev_err(platform->dev, + "ASoC: pcm constructor failed: %d\n", + ret); return ret; } } -- 1.7.10.4

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Re: [alsa-devel] [PATCH 1/2] ASoC: davinci-mcasp: Add pinctrl support
by Mark Brown 27 Nov '12

27 Nov '12

On Mon, Nov 26, 2012 at 09:55:33PM -0800, Peter Korsgaard wrote: > >>>>> "Mark" == Mark Brown <broonie(a)opensource.wolfsonmicro.com> writes: > Mark> Since the driver is happy to continue without the pins why is it > Mark> bothering requesting them in the first place? In other words, this > Mark> should be a hard error. > Sorry, I disagree. As was discussed recently, there's several reasons > why the pinctrl call might fail, and not all are fatal (E.G. pinmux > already setup by bootloader): > http://article.gmane.org/gmane.linux.kernel/1399756 I'd expect the platform to provide stub mappings for cases where the pinmux code shouldn't do anything.

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Re: [alsa-devel] [PATCH 2/2] ASoC: Davinci: machine: Add device tree binding
by Mark Brown 27 Nov '12

27 Nov '12

On Thu, Nov 22, 2012 at 09:34:59AM +0530, Hebbar, Gururaja wrote: > +- dai-data : A list of DAI data used by SOC code to register > + DAI, Codecs platform. > + The string index "should" be as shown below. > +dai-data = > +"<DAI Name>", "<DAI Stream Name>", > +"<CODEC DAI Name>", "<true if evm_aic3x_init is required, else false>", > +"<evm ops required (evm_ops or evm_spdif_ops)>"; This continues to have the same issue as with previous versions, this is all Linux internals (especially the evm_aic3x_init thing) not a description of the hardware, there's not enough information to allow someone to work out what the bindings for their system should be. Do not post *ANY* further versions that have this evm_aic3x_init stuff in them, it's very annoying that you keep on ignoring this feedback.

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