November 2013 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Commit:ce6cfaf1 query for shared control
by Lars-Peter Clausen 25 Nov '13

25 Nov '13

On 11/25/2013 08:12 AM, noman pouigt wrote: > Hello Lars, > > I was wondering if you can enlighten me > on below query. > > Commit:ce6cfaf1 > > From the commit text "The input and > output paths for each widgets are only > updated though during the respective run > for that widget". > > Does this commit text mean: when mux > or mixer_power_update is called then only > the widget is powered and it's peer- widget > source(path source) and widget sink(path sink)? Hi, What the commit does is to make sure that if a control is shared between multiple mixers/muxes to apply all the changes in the same update sequence. Instead of, as it was done before, to run a update sequence for each mixer/mux. > > +------+ > A1 ------| | > | MUX1 |----- C1 > B1 ------| | > +------+ > | > control ---+ > | > +------+ > A2 ------| | > | MUX2 |----- C2 > B2 ------| | > +------+ > > Can I represent the above diagram as below: > > char *input_text_1[] = { > "A1", "B1" > }; > > struct soc_enum input_enum_1 = > SOC_ENUM_SINGLE(some_register, some_bit, > 2, input_text_1); > > char *input_text_2[] = { > "A2", "B2" > }; > struct soc_enum input_enum_2 = > SOC_ENUM_SINGLE(some_register, some_bit, > 2, input_text_2); > > > struct snd_kcontrol_new input_mux = > SOC_DAPM_ENUM("C1", input_enum_1); > struct snd_kcontrol_new input_mux = > SOC_DAPM_ENUM("C2", input_enum_2); > > dapm_route[] { > "A1", "control","C1" > "B1", "control","C1" > "A2", "control","C2" > "B2", "control","C2" > }; The control part needs to have the name of one of the enum items. So this would be { "A1", "A1", "C1" } and so on. - Lars

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[alsa-devel] [PATCH] ASoC: dapm: Use SND_SOC_DAPM_INIT_REG_VAL in SND_SOC_DAPM_MUX
by Stephen Warren 24 Nov '13

24 Nov '13

From: Stephen Warren <swarren(a)nvidia.com> SND_SOC_DAPM_MUX() doesn't currently initialize the .mask field. This results in the mux never affecting HW, since no bits are ever set or cleared. Fix SND_SOC_DAPM_MUX() to use SND_SOC_DAPM_INIT_REG_VAL() to set up the reg, shift, on_val, and off_val fields like almost all other SND_SOC_xxx() macros. It looks like this was a "typo" in the fixed commit linked below. This makes the speakers on the Toshiba AC100 (PAZ00) laptop work again. Fixes: de9ba98b6d26 ("ASoC: dapm: Make widget power register settings more flexible") Cc: <stable(a)vger.kernel.org> # v3.12+ Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- include/sound/soc-dapm.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 2037c45adfe6..56ebdfca6273 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -104,7 +104,8 @@ struct device; SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .kcontrol_news = wcontrols, .num_kcontrols = 1} #define SND_SOC_DAPM_MUX(wname, wreg, wshift, winvert, wcontrols) \ -{ .id = snd_soc_dapm_mux, .name = wname, .reg = wreg, \ +{ .id = snd_soc_dapm_mux, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .kcontrol_news = wcontrols, .num_kcontrols = 1} #define SND_SOC_DAPM_VIRT_MUX(wname, wreg, wshift, winvert, wcontrols) \ { .id = snd_soc_dapm_virt_mux, .name = wname, \ -- 1.8.1.5

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[alsa-devel] [PATCH] ASoC: Rename mid-x86 directory to intel
by Jarkko Nikula 24 Nov '13

24 Nov '13

We have other Intel platforms coming having the Smart Sound Technology (SST) so rename the mid-x86 directory to intel as originally directory name reflected only Intel MID platform. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/Kconfig | 2 +- sound/soc/Makefile | 2 +- sound/soc/{mid-x86 => intel}/Kconfig | 0 sound/soc/{mid-x86 => intel}/Makefile | 0 sound/soc/{mid-x86 => intel}/mfld_machine.c | 0 sound/soc/{mid-x86 => intel}/sst_dsp.h | 0 sound/soc/{mid-x86 => intel}/sst_platform.c | 0 sound/soc/{mid-x86 => intel}/sst_platform.h | 0 8 files changed, 2 insertions(+), 2 deletions(-) rename sound/soc/{mid-x86 => intel}/Kconfig (100%) rename sound/soc/{mid-x86 => intel}/Makefile (100%) rename sound/soc/{mid-x86 => intel}/mfld_machine.c (100%) rename sound/soc/{mid-x86 => intel}/sst_dsp.h (100%) rename sound/soc/{mid-x86 => intel}/sst_platform.c (100%) rename sound/soc/{mid-x86 => intel}/sst_platform.h (100%) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 5138b8493051..463a9e25e04f 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -42,7 +42,7 @@ source "sound/soc/jz4740/Kconfig" source "sound/soc/nuc900/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/kirkwood/Kconfig" -source "sound/soc/mid-x86/Kconfig" +source "sound/soc/intel/Kconfig" source "sound/soc/mxs/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 8b9e70105dd2..ff291d3e60af 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -16,7 +16,7 @@ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += jz4740/ -obj-$(CONFIG_SND_SOC) += mid-x86/ +obj-$(CONFIG_SND_SOC) += intel/ obj-$(CONFIG_SND_SOC) += mxs/ obj-$(CONFIG_SND_SOC) += nuc900/ obj-$(CONFIG_SND_SOC) += omap/ diff --git a/sound/soc/mid-x86/Kconfig b/sound/soc/intel/Kconfig similarity index 100% rename from sound/soc/mid-x86/Kconfig rename to sound/soc/intel/Kconfig diff --git a/sound/soc/mid-x86/Makefile b/sound/soc/intel/Makefile similarity index 100% rename from sound/soc/mid-x86/Makefile rename to sound/soc/intel/Makefile diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/intel/mfld_machine.c similarity index 100% rename from sound/soc/mid-x86/mfld_machine.c rename to sound/soc/intel/mfld_machine.c diff --git a/sound/soc/mid-x86/sst_dsp.h b/sound/soc/intel/sst_dsp.h similarity index 100% rename from sound/soc/mid-x86/sst_dsp.h rename to sound/soc/intel/sst_dsp.h diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/intel/sst_platform.c similarity index 100% rename from sound/soc/mid-x86/sst_platform.c rename to sound/soc/intel/sst_platform.c diff --git a/sound/soc/mid-x86/sst_platform.h b/sound/soc/intel/sst_platform.h similarity index 100% rename from sound/soc/mid-x86/sst_platform.h rename to sound/soc/intel/sst_platform.h -- 1.8.4.2

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[alsa-devel] [patch] ASoC: rcar: some dubious one-bit signed bitfields
by Dan Carpenter 24 Nov '13

24 Nov '13

Because these are signed they can either be 0 or -1 instead of 0 and 1 as intended. It doesn't cause a problem from what I can see, but it's dangerous and Sparse complains: sound/soc/sh/rcar/rsnd.h:177:25: error: dubious one-bit signed bitfield Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9e463e5..b5ac3a2 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -174,11 +174,11 @@ struct rsnd_dai { struct rsnd_dai_stream playback; struct rsnd_dai_stream capture; - int clk_master:1; - int bit_clk_inv:1; - int frm_clk_inv:1; - int sys_delay:1; - int data_alignment:1; + unsigned int clk_master:1; + unsigned int bit_clk_inv:1; + unsigned int frm_clk_inv:1; + unsigned int sys_delay:1; + unsigned int data_alignment:1; }; #define rsnd_dai_nr(priv) ((priv)->dai_nr)

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[alsa-devel] [PATCH][RFC] ASoC: rcar: select REGMAP
by Kuninori Morimoto 24 Nov '13

24 Nov '13

55e5b6fd5af04b6d8b0ac6635edf49476ff298ba (ASoC: rsnd: use regmap instead of original register mapping method) support regmap/regmap_field on Renesas sound driver. It needs CONFIG_REGMAP now. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark I'm not sure this is needed or not, but I tried send it sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 14011d9..ff60e11 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD + select REGMAP help This option enables R-Car SUR/SCU/SSIU/SSI sound support -- 1.7.9.5

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[alsa-devel] [PATCH 0/4] *** SUBJECT HERE ***
by Charles Keepax 24 Nov '13

24 Nov '13

Hi, Small patch series adding some missing functionality on the WM5110. Given most of the changes are in ASoC I have generated the patches against Mark's tree but let me know if we want to push them through MFD instead and I can regenerate, but I suspect they will cause less conflicts going through ASoC. Thanks, Charles Charles Keepax (3): mfd: wm5110: Make DSP memories readable ASoC: wm5110: Add basic support for ISRCs ASoC: wm5110: Expose input high pass filter controls Mark Brown (1): ASoC: wm5110: Hook up ADSP2 cores drivers/mfd/wm5110-tables.c | 73 +++++++- include/linux/mfd/arizona/registers.h | 37 ++++ sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/arizona.c | 10 + sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/wm5110.c | 316 ++++++++++++++++++++++++++++++++- 6 files changed, 433 insertions(+), 6 deletions(-) -- 1.7.2.5

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Add missing __iomem annotations
by Lars-Peter Clausen 24 Nov '13

24 Nov '13

There are two places in the fsl_ssi driver where the pointer to the register set is missing the __iomem annotation. Adding it fixes the following warnings from sparse: sound/soc/fsl/fsl_ssi.c:747:45: warning: incorrect type in initializer (different address spaces) sound/soc/fsl/fsl_ssi.c:747:45: expected struct ccsr_ssi *ssi sound/soc/fsl/fsl_ssi.c:747:45: got struct ccsr_ssi [noderef] <asn:2>*ssi sound/soc/fsl/fsl_ssi.c:756:9: warning: incorrect type in argument 2 (different address spaces) sound/soc/fsl/fsl_ssi.c:756:9: expected void volatile [noderef] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:756:9: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:759:9: warning: incorrect type in argument 2 (different address spaces) sound/soc/fsl/fsl_ssi.c:759:9: expected void volatile [noderef] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:759:9: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:761:25: warning: incorrect type in argument 1 (different address spaces) sound/soc/fsl/fsl_ssi.c:761:25: expected unsigned int [noderef] [usertype] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:761:25: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:769:45: warning: incorrect type in initializer (different address spaces) sound/soc/fsl/fsl_ssi.c:769:45: expected struct ccsr_ssi *ssi sound/soc/fsl/fsl_ssi.c:769:45: got struct ccsr_ssi [noderef] <asn:2>*ssi sound/soc/fsl/fsl_ssi.c:775:9: warning: incorrect type in argument 2 (different address spaces) sound/soc/fsl/fsl_ssi.c:775:9: expected void volatile [noderef] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:775:9: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:776:25: warning: incorrect type in argument 1 (different address spaces) sound/soc/fsl/fsl_ssi.c:776:25: expected unsigned int [noderef] [usertype] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:776:25: got unsigned int *<noident> sound/soc/fsl/fsl_ssi.c:781:16: warning: incorrect type in argument 1 (different address spaces) sound/soc/fsl/fsl_ssi.c:781:16: expected void const volatile [noderef] <asn:2>*addr sound/soc/fsl/fsl_ssi.c:781:16: got unsigned int *<noident> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/fsl_ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 3df0318..1a9521a 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -751,7 +751,7 @@ static void fsl_ssi_ac97_init(void) static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val) { - struct ccsr_ssi *ssi = fsl_ac97_data->ssi; + struct ccsr_ssi __iomem *ssi = fsl_ac97_data->ssi; unsigned int lreg; unsigned int lval; @@ -773,7 +773,7 @@ static void fsl_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, static unsigned short fsl_ssi_ac97_read(struct snd_ac97 *ac97, unsigned short reg) { - struct ccsr_ssi *ssi = fsl_ac97_data->ssi; + struct ccsr_ssi __iomem *ssi = fsl_ac97_data->ssi; unsigned short val = -1; unsigned int lreg; -- 1.8.0

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[alsa-devel] Zoom R16
by Jason Mancine 24 Nov '13

24 Nov '13

Hello, I am looking for some assistance in getting the Zoom R16 working for playback. Please let me know if this is the right forum for this... The device is recognized, but I just can't figure out which interface is which, and how to configure them in quirks-table.h. Any help would be appreciated. To start, the output of lsusb -v for the device is this: --- Bus 001 Device 010: ID 1686:00dd ZOOM Corporation Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x1686 ZOOM Corporation idProduct 0x00dd bcdDevice 0.02 iManufacturer 1 iProduct 2 R16 iSerial 3 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 259 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 480mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 0b 24 01 00 01 35 00 03 01 02 03 ** UNRECOGNIZED: 0c 24 02 05 01 01 00 02 03 00 00 00 ** UNRECOGNIZED: 09 24 03 08 01 03 00 05 00 ** UNRECOGNIZED: 0c 24 02 09 01 02 00 08 00 00 00 00 ** UNRECOGNIZED: 09 24 03 0c 01 01 00 09 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 05 01 01 00 ** UNRECOGNIZED: 14 24 02 01 02 04 18 04 44 ac 00 80 bb 00 88 58 01 00 77 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x006c 1x 108 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 0c 01 01 00 ** UNRECOGNIZED: 14 24 02 01 08 04 18 04 44 ac 00 80 bb 00 88 58 01 00 77 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x01a0 1x 416 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 07 24 01 00 01 41 00 ** UNRECOGNIZED: 06 24 02 01 01 00 ** UNRECOGNIZED: 06 24 02 02 02 00 ** UNRECOGNIZED: 09 24 03 01 03 01 02 01 00 ** UNRECOGNIZED: 09 24 03 02 04 01 01 01 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 0 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 bNumConfigurations 1 Device Status: 0x0001 Self Powered --- This is the quirk that Clemens provided on the linux-audio mail list...the device is recognized, but no capture/playback: { /* ZOOM R16 in USB 2.0 mode */ USB_DEVICE(0x1686, 0x00dd), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 3, .type = QUIRK_MIDI_STANDARD_INTERFACE }, { .ifnum = .1 }, } } }, Now....despite there being no #4 interface in lsusb, if I append the following to the above quirk I get all 8 channels of capture working. I am more confused than ever. :) { .ifnum = 4, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_LE, .channels = 8, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x84, .ep_attr = 13, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, .rate_min = 44100, .rate_max = 96000, .nr_rates = 4, .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 } } }, Obviously, this is not the correct solution and I can't make playback work by adding additional .ifnums. I am quite certain that interface 1 is capture and 2 is playback but if I add the same info to either of those sections my entire system freezes when I plug in the device. Thanks, jason

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Re: [alsa-devel] [pulseaudio-discuss] acceptable noise level for mic boost
by Oleksij Rempel 23 Nov '13

23 Nov '13

Am 22.11.2013 18:34, schrieb Oleksij Rempel: > Am 22.11.2013 17:48, schrieb Tanu Kaskinen: >> On Thu, 2013-11-21 at 16:42 +0100, Oleksij Rempel wrote: >>> Hello all, >>> >>> i posted this email to ALSA, but probably PA devs have some ideas too. >>> >>> ====================================================================== >>> Hello Takashi, >>> >>> what is acceptable noise level for mic boost? >>> My laptop has three levels of mic boost (+36dB). Level 3 is useless >>> since noise level will be about 0db. With level 1 i get -12dB noise, so >>> there is almost 12dB room for data... but it is not enough for normal >>> speech, which is about 20dB. >>> >>> Are there any ALSA guide for HW designers, which will say some thing >>> like acceptable levels are 90-20dB? >>> There are some Skype and Microsoft prescriptions for certification. >>> For example M$ prescribe 18dB or more for speech to noise ration for >>> build in mics. It sound like good number to start. >>> >>> In my case, i need to remove mic boost completely to guarantee minimum >>> 18dB SpNR. Suddenly i didn't found correct way to do it. >>> I will be thankful for some tip. >>> >>> ===================================================================== >>> >>> So my question to PA devs. Do pulse audio need to know about noise level >>> of microphone? Will it help to make correct decisions? Especial for VoIP >>> and automatic gain control. >> >> We don't currently have any idea of the noise level, so we don't have >> code for adjusting our decisions based on that information either. So, >> I'd say that pulseaudio doesn't *need* to know about the noise level. Of >> course, if someone someday complains that pulseaudio does bad decisions >> because it doesn't take the noise level into account, then we would need >> that information. >> > > Thank you for your response. > I have more questions: > Skype continuously trying to adjust record level. If there is a silence > it will try to push the input volume as close to 0dB (in pulseaudio > terms) as possible. > For my laptop 0dB(in PA) will mean some thing +68dB over base volume > (ALSA). The noise will reach 0dBFS. > Well, it wont kill any one, but it sound bad. > > What 0dB mean in pulseaudio? > Why pulseaudio has capture volume range -130 - +11dB? Is it possible to > fit this gain to 16-bit PCM? to answer this question by my self. If i understand correctly pulseaudio trying to imitate professional mixer settings. The difference is, usual mixer has 0dB settings as point for preconfigured sound. You need to use "Gain" or "Trim" control to configure it to acceptable level. 0dB in PulseAudio looks for me incorrect, if no preconfigurable available. Here is one example: http://www.youtube.com/watch?v=YXSJ1oOQ9tY > Now to my hardware. I can blacklist in the driver 2 "mic boost" steps > without introducing any regressions. Which will guaranty maximal noise > level = -12dBFS. But it is still not optimal. Best experience i had with > limit to -24dB. In this case i need to remove Mic Boost completely but > pulseaudio will fail to make a difference between internal and external mic. > > What is best solution in this case? Fix pulse or leave at least one step > for mic boost? After some more testing, i'm sure it should be fixed in pulseaudio. Blacklisting microfone setting is just quick dirty hack. It will work for internal mic. But fail on external. There is no way, ALSA can predict some thing about it. The best thing ALSA can do here is to provide all possible controls. > The microphone on this laptop is a digital mic. Do it make sense to use > hardware amp? Or is it possible to get better results with software? Theoretically, Skype or Empathy or any other app with auto gain control should us 0dB as target. But since ALSA or PulseAudio can't automatically detect correct point, PuleAudio should provide possibility to override 0dB offset. For build in microphones can be used ALSA quirk list or some kind of database for pulseaudio with per device default values to make better default experience. IMO, current solution with blacklists in ALSA is wrong. -- Regards, Oleksij

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[alsa-devel] [PATCH] ASoC: ak4641: Convert to direct regmap API usage
by Mark Brown 22 Nov '13

22 Nov '13

From: Mark Brown <broonie(a)linaro.org> We're trying to remove the ASoC level I/O functions. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/ak4641.c | 38 +++++++++++++++++++++++++------------- 1 file changed, 25 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 49cc5f6d6dba..94cbe508dd37 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -17,6 +17,7 @@ #include <linux/gpio.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -30,6 +31,7 @@ /* codec private data */ struct ak4641_priv { + struct regmap *regmap; unsigned int sysclk; int deemph; int playback_fs; @@ -38,12 +40,12 @@ struct ak4641_priv { /* * ak4641 register cache */ -static const u8 ak4641_reg[AK4641_CACHEREGNUM] = { - 0x00, 0x80, 0x00, 0x80, - 0x02, 0x00, 0x11, 0x05, - 0x00, 0x00, 0x36, 0x10, - 0x00, 0x00, 0x57, 0x00, - 0x88, 0x88, 0x08, 0x08 +static const struct reg_default ak4641_reg_defaults[] = { + { 0, 0x00 }, { 1, 0x80 }, { 2, 0x00 }, { 3, 0x80 }, + { 4, 0x02 }, { 5, 0x00 }, { 6, 0x11 }, { 7, 0x05 }, + { 8, 0x00 }, { 9, 0x00 }, { 10, 0x36 }, { 11, 0x10 }, + { 12, 0x00 }, { 13, 0x00 }, { 14, 0x57 }, { 15, 0x00 }, + { 16, 0x88 }, { 17, 0x88 }, { 18, 0x08 }, { 19, 0x08 } }; static const int deemph_settings[] = {44100, 0, 48000, 32000}; @@ -396,6 +398,7 @@ static int ak4641_mute(struct snd_soc_dai *dai, int mute) static int ak4641_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct ak4641_priv *ak4641 = snd_soc_codec_get_drvdata(codec); struct ak4641_platform_data *pdata = codec->dev->platform_data; int ret; @@ -417,7 +420,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, gpio_set_value(pdata->gpio_npdn, 1); mdelay(1); - ret = snd_soc_cache_sync(codec); + ret = regcache_sync(ak4641->regmap); if (ret) { dev_err(codec->dev, "Failed to sync cache: %d\n", ret); @@ -433,7 +436,7 @@ static int ak4641_set_bias_level(struct snd_soc_codec *codec, gpio_set_value(pdata->gpio_npdn, 0); if (pdata && gpio_is_valid(pdata->gpio_power)) gpio_set_value(pdata->gpio_power, 0); - codec->cache_sync = 1; + regcache_mark_dirty(ak4641->regmap); break; } codec->dapm.bias_level = level; @@ -518,7 +521,7 @@ static int ak4641_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -550,12 +553,17 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4641 = { .dapm_routes = ak4641_audio_map, .num_dapm_routes = ARRAY_SIZE(ak4641_audio_map), .set_bias_level = ak4641_set_bias_level, - .reg_cache_size = ARRAY_SIZE(ak4641_reg), - .reg_word_size = sizeof(u8), - .reg_cache_default = ak4641_reg, - .reg_cache_step = 1, }; +static const struct regmap_config ak4641_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK4641_BTIF, + .reg_defaults = ak4641_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4641_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; static int ak4641_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) @@ -569,6 +577,10 @@ static int ak4641_i2c_probe(struct i2c_client *i2c, if (!ak4641) return -ENOMEM; + ak4641->regmap = devm_regmap_init_i2c(i2c, &ak4641_regmap); + if (IS_ERR(ak4641->regmap)) + return PTR_ERR(ak4641->regmap); + if (pdata) { if (gpio_is_valid(pdata->gpio_power)) { ret = gpio_request_one(pdata->gpio_power, -- 1.8.4.4

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