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

[alsa-devel] mint 13 mate + ATI IXP SB4x0 High Definition Audio Controller (rev 01)
by Pavel 20 Jun '13

20 Jun '13

Hello! i have a big problem - no sound after install linux mint 13 mate. well, sometimes if i reboot - i have it, but after nex reboot - no sound no more, seems like pulseaudio don't see my sound card. Laptop: Manufacturer: FUJITSU SIEMENS Product Name: AMILO Pa 1510 Product Version: Not Applicable Firmware Version: 1.07C soundcard: 00:14.2 Audio device: Advanced Micro Devices [AMD] nee ATI IXP SB4x0 High Definition Audio Controller (rev 01) kernel 3.2.0-48-generic dmesg: http://pastebin.com/KF8ABwnY alsa-info: http://pastebin.com/QYBYNJi9 i read this topic: https://bugzilla.kernel.org/show_bug.cgi?id=18512 and tried to add "options snd-hda-intel model=6stack-dig single_cmd=1" to alsa-base.conf - it helped, after reboot i has sound, till i reboot - now no sound no more.. i tried everything i found in the google, even alsa-update - http://www.stchman.com/alsa_update.html nothing can help me, but i understand that's the problem is in the driver.. can someone pls help me to fix the problem? thank you.

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[alsa-devel] mint 13 mate + ATI IXP SB4x0 High Definition Audio Controller (rev 01)
by Pavel 20 Jun '13

20 Jun '13

Hello! i have a big problem - no sound after install linux mint 13 mate. well, sometimes if i reboot - i have it, but after nex reboot - no sound no more, seems like pulseaudio don't see my sound card. Laptop: Manufacturer: FUJITSU SIEMENS Product Name: AMILO Pa 1510 Product Version: Not Applicable Firmware Version: 1.07C soundcard: 00:14.2 Audio device: Advanced Micro Devices [AMD] nee ATI IXP SB4x0 High Definition Audio Controller (rev 01) kernel 3.2.0-48-generic dmesg: http://pastebin.com/KF8ABwnY alsa-info: http://pastebin.com/QYBYNJi9 i read this topic: https://bugzilla.kernel.org/show_bug.cgi?id=18512 and tried to add "options snd-hda-intel model=6stack-dig single_cmd=1" to alsa-base.conf - it helped, after reboot i has sound, till i reboot - now no sound no more.. i tried everything i found in the google, even alsa-update - http://www.stchman.com/alsa_update.html nothing can help me, but i understand that's the problem is in the driver.. can someone pls help me to fix the problem? thank you.

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[alsa-devel] cs42l52 alsa driver broken beep volume control
by Lars-Peter Clausen 20 Jun '13

20 Jun '13

Hi, I stumbled upon this while making some changes to the SOC_SINGLE_VALUE macro. For the Beep Volume control the cs42l52 driver sets the invert field to 7, which doesn't make much sense and is causing a build warning with my changes. I assume the intention was to actually use SOC_SINGLE_SX_TLV instead of SOC_SINGLE_TLV. But I'm not sure if setting min to 0x1f and max 0x7 is right either. Also according to the datasheet the tlv dB info seems to be wrong. The Beep Volume uses 2dB steps while the info uses 0.5dB steps. - Lars

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[alsa-devel] [PATCH v8 00/11] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 20 Jun '13

20 Jun '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. In version 8 I fixed the build issues by seperating the DT capable ac97 phycore driver from the old driver. Regards, Markus Changes in v8: - Fix build issues in phycore-ac97. Changes in v7: - Only one soc_ac97_ops struct for both ssi drivers (fsl-ssi and imx-ssi) to make it possible to compile a kernel with both drivers. - Other small bug/style fixes described in the changelogs of the patches. Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (11): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move soc_ac97_ops from imx-ssi to fsl_ssi ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: imx ac97 support ARM: imx: Export ac97 reset functions ASoC: Add phycore-ac97-dt driver ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 20 +- sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 450 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 15 +- sound/soc/fsl/imx-ssi.c | 14 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97-dt.c | 243 +++++++++++ sound/soc/fsl/phycore-ac97.c | 13 +- 15 files changed, 706 insertions(+), 116 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt create mode 100644 sound/soc/fsl/phycore-ac97-dt.c

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[alsa-devel] [PATCH] ASoC: SPEAr: Add device tree support + SPEAr machine driver
by Rajeev Kumar 20 Jun '13

20 Jun '13

The patch adds device tree support + SPEAr machine driver Signed-off-by: Rajeev Kumar <rajeev-dlh.kumar(a)st.com> --- .../devicetree/bindings/sound/spear-evb.txt | 20 +++ sound/soc/spear/spear_evb.c | 135 ++++++++++++++++++++ 2 files changed, 155 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/spear-evb.txt create mode 100644 sound/soc/spear/spear_evb.c diff --git a/Documentation/devicetree/bindings/sound/spear-evb.txt b/Documentation/devicetree/bindings/sound/spear-evb.txt new file mode 100644 index 0000000..cb011b4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/spear-evb.txt @@ -0,0 +1,20 @@ +* SPEAr audio sta529 codec + +Required properties: +- compatible: "spear,spear-evb" +- audio-controllers: The phandle list of synopsys audio i2s controller +- audio-codec: The phandle of the sta529 audio codec +- codec_dai_name: DAI name for codec. +- stream_name: stream name. + +Example: + +sound { + compatible = "spear,spear-evb"; + audio-controllers = <&i2s0 &i2s1>; + audio-codecs = <&sta529 &sta529>; + codec_dai_name = "sta529-audio", "sta529-audio"; + stream_name = "i2s-play", "i2s-cap"; + dai_name = "i2s0-pcm", "i2s1-pcm"; +}; + diff --git a/sound/soc/spear/spear_evb.c b/sound/soc/spear/spear_evb.c new file mode 100644 index 0000000..6da1732 --- /dev/null +++ b/sound/soc/spear/spear_evb.c @@ -0,0 +1,135 @@ +/* + * ASoC machine driver for SPEAr evaluation boards + * + * sound/soc/spear/spear_evb.c + * + * Copyright (C) 2010 ST Microelectronics + * Rajeev Kumar<rajeev-dlh.kumar(a)st.com> + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include <linux/io.h> +#include <linux/of.h> +#include <linux/platform_device.h> +#include <linux/module.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/spear_dma.h> + +struct snd_soc_card spear_soc_card; + +static int spear_evb_probe(struct platform_device *pdev) +{ + struct snd_soc_dai_link *evb_dai; + struct device_node *np = pdev->dev.of_node; + struct device_node *spear_audio_np, *codec_np; + int i, nr_controllers, ret = 0; + + if (!np) + return -EINVAL; /* no device tree */ + + of_property_read_u32(np, "nr_controllers", &nr_controllers); + + evb_dai = devm_kzalloc(&pdev->dev, + (nr_controllers * sizeof(*evb_dai)), GFP_KERNEL); + if (!evb_dai) { + dev_warn(&pdev->dev, "kzalloc fail for evb_dai\n"); + return -ENOMEM; + } + + for (i = 0; i < nr_controllers ; i++) { + spear_audio_np = of_parse_phandle(np, "audio-controllers", i); + codec_np = of_parse_phandle(np, "audio-codecs", i); + if (!spear_audio_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + return -EINVAL; + } + + ret = of_property_read_string_index(np, "dai_names", i, + &evb_dai[i].name); + if (ret < 0) { + dev_err(&pdev->dev, "Cannot parse names: %d\n", ret); + goto err; + } + + ret = of_property_read_string_index(np, "codec_dai_names", i, + &evb_dai[i].codec_dai_name); + if (ret < 0) { + dev_err(&pdev->dev, "Can't parse codec-dai-names: %d\n", + ret); + goto err; + } + ret = of_property_read_string_index(np, "stream_names", i, + &evb_dai[i].stream_name); + if (ret < 0) { + dev_err(&pdev->dev, "Cannot parse stream names: %d\n", + ret); + goto err; + } + + evb_dai[i].cpu_dai_name = NULL; + evb_dai[i].cpu_of_node = spear_audio_np; + evb_dai[i].platform_of_node = np; + evb_dai[i].codec_name = NULL; + evb_dai[i].codec_of_node = codec_np; + if (!(strcmp(evb_dai[i].codec_dai_name, "sta529-audio"))) + evb_dai[i].dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBS_CFM; + of_node_put(spear_audio_np); + of_node_put(codec_np); + } + + spear_soc_card.name = "spear-evb"; + spear_soc_card.dai_link = evb_dai; + spear_soc_card.num_links = nr_controllers; + + spear_soc_card.dev = &pdev->dev; + platform_set_drvdata(pdev, &spear_soc_card); + + ret = snd_soc_register_card(&spear_soc_card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static int spear_evb_remove(struct platform_device *pdev) +{ + struct snd_soc_card *spear_soc_card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(spear_soc_card); + + return 0; +} + +static const struct of_device_id spear_evb_dt_ids[] = { + { .compatible = "spear,spear-evb", }, + { } +}; +MODULE_DEVICE_TABLE(of, spear_evb_dt_ids); + +static struct platform_driver spear_evb_driver = { + .driver = { + .name = "spear-evb", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = spear_evb_dt_ids, + }, + .probe = spear_evb_probe, + .remove = spear_evb_remove, +}; + +module_platform_driver(spear_evb_driver); + +MODULE_AUTHOR("Rajeev Kumar <rajeev-dlh.kumar(a)st.com>"); +MODULE_DESCRIPTION("ST SPEAr EVB ASoC driver"); +MODULE_LICENSE("GPL"); -- 1.6.0.2

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[alsa-devel] [PATCH] sound: using strlcpy instead of strncpy
by Zhao Hongjiang 20 Jun '13

20 Jun '13

for NUL terminated string, need alway set '\0' in the end. Signed-off-by: Zhao Hongjiang <zhaohongjiang(a)huawei.com> --- sound/oss/soundcard.c | 2 +- sound/soc/codecs/88pm860x-codec.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index e778034..53ac987 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -99,7 +99,7 @@ int *load_mixer_volumes(char *name, int *levels, int present) } n = num_mixer_volumes++; - strncpy(mixer_vols[n].name, name, 32); + strlcpy(mixer_vols[n].name, name, sizeof(mixer_vols[n].name)); if (present) mixer_vols[n].num = n; diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 60159c0..6b2e6b3 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1447,7 +1447,7 @@ static int pm860x_codec_probe(struct platform_device *pdev) goto out; } pm860x->irq[i] = res->start + chip->irq_base; - strncpy(pm860x->name[i], res->name, MAX_NAME_LEN); + strlcpy(pm860x->name[i], res->name, sizeof(pm860x->name[i])); } ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pm860x, -- 1.7.1

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[alsa-devel] [PATCH] ALSA: hda - Make Thinkpad X220-tablet use generic parser
by David Henningsson 20 Jun '13

20 Jun '13

Like the X220, this quirk was added to support docking station, so enable the fixup instead. According to Jan, the generic parser works equal or better than the current parser. This was tested under a 3.9 kernel. Reported-by: Jan Alexander Steffens <jan.steffens(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b314d3e..de00ce1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2947,7 +2947,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c5, "Thinkpad Edge 13", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21c6, "Thinkpad Edge 13", CXT5066_ASUS), - SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), {} @@ -3318,6 +3317,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21da, "Lenovo X220", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21db, "Lenovo X220-tablet", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), -- 1.7.9.5

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[alsa-devel] [alsa-ucm] segfault at __list_del function in list.h
by sajeesh sidharthan 20 Jun '13

20 Jun '13

Hello. I am testing pulseaudio ucm. I have faced a crash when active ucm device list empty. Please let me know if following fix can be checked in alsa-lib-1.0.25/include/list.h old code: static __inline__ void list_del(struct list_head *entry) { __list_del(entry->prev, entry->next); } new code: static __inline__ void list_del(struct list_head *entry) { if(entry->prev && entry->next) __list_del(entry->prev, entry->next); } Sajeesh

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Re: [alsa-devel] [RFC PATCH 00/11] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Mark Brown 20 Jun '13

20 Jun '13

On Wed, Jun 19, 2013 at 09:32:44PM +0200, Tomasz Figa wrote: > On Wednesday 19 of June 2013 20:22:11 Mark Brown wrote: > > No, I didn't - that's most likely it, I didn't really investigate. I > > didn't test the watchdog stuff as the clocks didn't get sent to me. > I always try to keep you on Cc of my patches for s3c64xx, as you are the > most active user of this platform (if not the only one other than me) and > this was the case for clock patches as well, just checked that. Yes, I had the clock patches but they depended on something or other I think. I did test a previous verion IIRC. There are other users I'm aware of (you can probably take a wild guess where from the board) though they tend not to be terribly active upstream.

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Re: [alsa-devel] [RFC PATCH 11/11] ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver
by Linus Walleij 19 Jun '13

19 Jun '13

On Wed, Jun 19, 2013 at 8:23 PM, Tomasz Figa <tomasz.figa(a)gmail.com> wrote: > On Monday 17 of June 2013 16:04:47 Linus Walleij wrote: >> On Sun, Jun 16, 2013 at 10:54 PM, Tomasz Figa <tomasz.figa(a)gmail.com> > wrote: >> > +static AMBA_AHB_DEVICE(s3c64xx_dma1, "dma-pl080s.1", 0x0a141080, >> > + 0x75100000, {IRQ_DMA1}, >> > &s3c64xx_dma1_plat_data); >> Ah, does this mean the device does not have any AMBA PrimeCell ID >> contents? > > Hmm, not really. The ID here is what I copied from PrimeCell ID registers > and the method of registration is taken from S5PV210. If I remove the > override, it will read the same value from registers. OK then don't override. Overriding is for hardware which is all zeroes in these registers. >> Then use the ID 0x00053080 instead of 0x0a141080 >> >> 0x53 is "S" for Samsung. Variant 0. > > Well, that's kind of solution to make sure that it won't conflict with any > future variant manufactured by ARM, I guess. Yeah, but as stated, if the hardware has proper IDs (these are probably inspired by the PCI vendor/device/revision fields) then rely on those. Yours, Linus Walleij

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Re: [alsa-devel] [RFC PATCH 00/11] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Arnd Bergmann 19 Jun '13

19 Jun '13

On Wednesday 19 June 2013, Tomasz Figa wrote: > On Wednesday 19 of June 2013 18:40:47 Mark Brown wrote: > > On Sun, Jun 16, 2013 at 10:54:07PM +0200, Tomasz Figa wrote: > > > One of the biggest roadblocks on the way of S3C64xx to DeviceTree > > > support is its DMA driver, which is completely platform-specific and > > > provides private API (s3c-dma), not even saying that its design is > > > completely against multiplatform-awareness. > > > > I tried to test this on my s3c64xx based system but it gave me a kernel > > that didn't boot far enough to give console output (there's some early > > init stuff that uses SPI...). That said, I needed: > > > > diff --git a/drivers/dma/amba-pl08x.c b/drivers/dma/amba-pl08x.c > > index 210a893..0f49707 100644 > > --- a/drivers/dma/amba-pl08x.c > > +++ b/drivers/dma/amba-pl08x.c > > @@ -313,7 +313,7 @@ static int pl08x_request_mux(struct pl08x_dma_chan > > *plchan) int ret; > > > > if (plchan->mux_use++ == 0 && pd->get_signal) { > > - ret = pd->get_signal(plchan->cd); > > + ret = (pd->get_signal)(plchan->cd); > > Hmm, that's strange. The former is a completely valid piece of code... get_signal is a macro defined in include/linux/signal.h. If that header gets included, neither of the two is valid. Arnd

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[alsa-devel] [PATCH 01/11] ASoC: tlv320aix3x: Use SOC_SINGLE_EXT() instead of open-coding it
by Lars-Peter Clausen 19 Jun '13

19 Jun '13

Cc: Jarkko Nikula <jhnikula(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/tlv320aic3x.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 1514bf8..e5b9268 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -128,10 +128,8 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_soc_info_volsw, \ - .get = snd_soc_dapm_get_volsw, .put = snd_soc_dapm_put_volsw_aic3x, \ - .private_value = SOC_SINGLE_VALUE(reg, shift, mask, invert) } + SOC_SINGLE_EXT(xname, reg, shift, mask, invert, \ + snd_soc_dapm_get_volsw, snd_soc_dapm_put_volsw_aic3x) /* * All input lines are connected when !0xf and disconnected with 0xf bit field, -- 1.8.0

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[alsa-devel] [RFC PATCH 00/11] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 19 Jun '13

19 Jun '13

One of the biggest roadblocks on the way of S3C64xx to DeviceTree support is its DMA driver, which is completely platform-specific and provides private API (s3c-dma), not even saying that its design is completely against multiplatform-awareness. The DMA controller present on this SoC series is a custom variant of ARM PrimeCell PL080 modified by Samsung to add some extra features. It is mostly compatible with original PL080, except: - CH_CONTROL2 register is added between CH_CONTROL and CH_CONFIG, - offset of CH_CONFIG register is different, - transfer size field is moved from CH_CONTROL to CH_CONTROL2, - transfer size field is extended to 24 bits, allowing much bigger single transfer, - LLI consists of one more word, to account for CH_CONTROL2 register. Since all the rest is fully compatible with standard PL080 there is no point in having separate driver just for this single variant, so I decided to look into adding support for it to the amba-pl08x driver. There was already some attempt to achieve this before, but this was before Russel's big rework of the driver to use virtual channels, making the old patches being not much of use. This RFC series is a proof of concept that I managed to make during last days of hacking. Except one patch adding clkdev lookup to clock driver (which is being replaced with a CCF-compliant driver ATM), this is enough to get memcpy and slave transfers to work on S3C64xx. I have tested this on Mini6410 and SMDK6410 boards using dmatest for memcpy and Samsung I2S with madplay/aplay for slave transfers. Unfortunately I do not have access to other platforms with PL08x so I could not test for any regressions introduced on them. Credits for two patches go to Alban Bedel, who made a series fixing this driver to make it usable with audio drivers. I rebased his patches on top of mine and corrected coding style a bit. OK, that's all. Any comments are welcome. Feel free to start throwing eggs and tomatoes if you find this awful, but I won't be upset if I get some Tested-by or Acked-by as well. ;) Alban Bedel (2): dmaengine: PL08x: Fix reading the byte count in cctl dmaengine: PL08x: Add cyclic transfer support Tomasz Figa (9): dma: amba-pl08x: Use bitmap to pass variant specific quirks dma: amba-pl08x: Refactor pl08x_getbytes_chan() to lower indentation dma: amba-pl08x: Add support for different offset of CONFIG register dma: amba-pl08x: Add support for PL080S variant dma: amba-pl08x: Add support for different maximum transfer size dma: amba-pl08x: Keep LLIs aligned to 4-word boundary spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 8 +- arch/arm/mach-s3c64xx/Makefile | 1 + arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/include/mach/dma.h | 65 +++++ arch/arm/mach-s3c64xx/pl080.c | 244 +++++++++++++++++++ drivers/dma/amba-pl08x.c | 399 ++++++++++++++++++++++--------- drivers/spi/Kconfig | 2 +- include/linux/amba/pl080.h | 1 + sound/soc/samsung/Kconfig | 2 +- 10 files changed, 613 insertions(+), 115 deletions(-) create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.2.1

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Re: [alsa-devel] [PATCH v3] sound/sgtl5000: fix codec register initial values and handling
by Mark Brown 19 Jun '13

19 Jun '13

On Wed, Jun 19, 2013 at 01:16:06PM +0000, Oskar Schirmer wrote: This is a set of three unrelated changes, they should be sent as three separate patches. > Further, mask computation in declarative part is obviously wrong: > Fix FRAC DIVISOR to provide an 11 bit mask correctly. What makes you say that this is obviously wrong? It's not entirely clear. > -#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 > +#define SGTL5000_PLL_FRAC_DIV_MASK 0x07ff Based on your changelog I was expecting a mask with only two bits set here?

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Re: [alsa-devel] [PATCH] ALSA: firewire: fix error return code in scs_probe()
by Takashi Iwai 18 Jun '13

18 Jun '13

At Tue, 18 Jun 2013 21:09:42 +0800, Wei Yongjun wrote: > > From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> > > Fix to return -ENOMEM in the kmalloc() error handling > case instead of 0, as done elsewhere in this function. > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Thanks, applied with ack by Clemens. Takashi > --- > sound/firewire/scs1x.c | 4 +++- > 1 file changed, 3 insertions(+), 1 deletion(-) > > diff --git a/sound/firewire/scs1x.c b/sound/firewire/scs1x.c > index 844a555..b252c21 100644 > --- a/sound/firewire/scs1x.c > +++ b/sound/firewire/scs1x.c > @@ -405,8 +405,10 @@ static int scs_probe(struct device *unit_dev) > scs->output_idle = true; > > scs->buffer = kmalloc(HSS1394_MAX_PACKET_SIZE, GFP_KERNEL); > - if (!scs->buffer) > + if (!scs->buffer) { > + err = -ENOMEM; > goto err_card; > + } > > scs->hss_handler.length = HSS1394_MAX_PACKET_SIZE; > scs->hss_handler.address_callback = handle_hss; >

1 0

[alsa-devel] [PATCH V1] ALSA: hda - Avoid choose same converter for unused pins
by Wang Xingchao 18 Jun '13

18 Jun '13

For Intel Haswell HDMI codecs, the pins choose converter 0 by default. This would cause conflict when playing audio on unused pins,the pin with physical device connected would get audio data too. i.e. Pin 0/1/2 default choose converter 0, pin 1 has HDMI monitor connected. when play audio on Pin 0 or pin 2, pin 1 could get audio data too. This patch configure unused pins to choose different converter. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 91 ++++++++++++++++++++++++++++++++++++-------- 1 file changed, 76 insertions(+), 15 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8983747..43f8b76 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1110,26 +1110,15 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return 0; } -/* - * HDA PCM callbacks - */ -static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int hdmi_choose_cvt(struct hda_codec *codec, + int pin_idx, int *cvt_id, int *mux_id) { struct hdmi_spec *spec = codec->spec; - struct snd_pcm_runtime *runtime = substream->runtime; - int pin_idx, cvt_idx, mux_idx = 0; struct hdmi_spec_per_pin *per_pin; - struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; + int cvt_idx, mux_idx = 0; - /* Validate hinfo */ - pin_idx = hinfo_to_pin_index(spec, hinfo); - if (snd_BUG_ON(pin_idx < 0)) - return -EINVAL; per_pin = get_pin(spec, pin_idx); - eld = &per_pin->sink_eld; /* Dynamically assign converter to stream */ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { @@ -1147,17 +1136,89 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, continue; break; } + /* No free converters */ if (cvt_idx == spec->num_cvts) return -ENODEV; + if (cvt_id) + *cvt_id = cvt_idx; + if (mux_id) + *mux_id = mux_idx; + + return 0; +} + +static void haswell_config_cvts(struct hda_codec *codec, + int pin_id, int mux_id) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin; + int pin_idx, mux_idx; + int curr; + int err; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + per_pin = get_pin(spec, pin_idx); + + if (pin_idx == pin_id) + continue; + + curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + + /* Choose another unused converter */ + if (curr == mux_id) { + err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); + if (err < 0) + return; + snd_printdd("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + mux_idx); + } + } +} + +/* + * HDA PCM callbacks + */ +static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int pin_idx, cvt_idx, mux_idx = 0; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + struct hdmi_spec_per_cvt *per_cvt = NULL; + int err; + + /* Validate hinfo */ + pin_idx = hinfo_to_pin_index(spec, hinfo); + if (snd_BUG_ON(pin_idx < 0)) + return -EINVAL; + per_pin = get_pin(spec, pin_idx); + eld = &per_pin->sink_eld; + + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx); + if (err < 0) + return err; + + per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; - snd_hda_codec_write(codec, per_pin->pin_nid, 0, + snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); + + /* configure unused pins to choose other converters */ + if (codec->vendor_id == 0x80862807) + haswell_config_cvts(codec, pin_idx, mux_idx); + snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ -- 1.7.9.5

3 2

Re: [alsa-devel] [PATCH] ALSA: firewire: fix error return code in scs_probe()
by Clemens Ladisch 18 Jun '13

18 Jun '13

Wei Yongjun wrote: > Fix to return -ENOMEM in the kmalloc() error handling > case instead of 0, as done elsewhere in this function. > > Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Acked-by: Clemens Ladisch <clemens(a)ladisch.de> > scs->buffer = kmalloc(HSS1394_MAX_PACKET_SIZE, GFP_KERNEL); > - if (!scs->buffer) > + if (!scs->buffer) { > + err = -ENOMEM; > goto err_card; > + }

1 0

[alsa-devel] [PATCH] ALSA: hda - Avoid choose same converter for unused pins
by Wang Xingchao 18 Jun '13

18 Jun '13

For Intel Haswell HDMI codecs, the pins choose converter 0 by default. This would cause conflict when playing audio on unused pins,the pin with physical device connected would get audio data too. i.e. Pin 0/1/2 default choose converter 0, pin 1 has HDMI monitor connected. when play audio on Pin 0 or pin 2, pin 1 could get audio data too. This patch configure unused pins to choose different converter. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 89 +++++++++++++++++++++++++++++++++++++------- 1 file changed, 75 insertions(+), 14 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8983747..e887584 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1110,26 +1110,15 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, return 0; } -/* - * HDA PCM callbacks - */ -static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) +static int hdmi_choose_cvt(struct hda_codec *codec, + int pin_idx, int *cvt_id, int *mux_id) { struct hdmi_spec *spec = codec->spec; - struct snd_pcm_runtime *runtime = substream->runtime; - int pin_idx, cvt_idx, mux_idx = 0; struct hdmi_spec_per_pin *per_pin; - struct hdmi_eld *eld; struct hdmi_spec_per_cvt *per_cvt = NULL; + int cvt_idx, mux_idx = 0; - /* Validate hinfo */ - pin_idx = hinfo_to_pin_index(spec, hinfo); - if (snd_BUG_ON(pin_idx < 0)) - return -EINVAL; per_pin = get_pin(spec, pin_idx); - eld = &per_pin->sink_eld; /* Dynamically assign converter to stream */ for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { @@ -1147,10 +1136,77 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, continue; break; } + /* No free converters */ if (cvt_idx == spec->num_cvts) return -ENODEV; + if (cvt_id) + *cvt_id = cvt_idx; + if (mux_id) + *mux_id = mux_idx; + + return 0; +} + +static void haswell_config_cvts(struct hda_codec *codec, + int pin_id, int mux_id) +{ + struct hdmi_spec *spec = codec->spec; + struct hdmi_spec_per_pin *per_pin; + int pin_idx, mux_idx; + int curr = -1; + int err; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + per_pin = get_pin(spec, pin_idx); + + if (pin_idx == pin_id) + continue; + + curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + AC_VERB_GET_CONNECT_SEL, 0); + + /* Choose another unused converter */ + if (curr == mux_id) { + err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); + if (err < 0) + return; + snd_printk("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); + snd_hda_codec_write(codec, per_pin->pin_nid, 0, + AC_VERB_SET_CONNECT_SEL, + mux_idx); + } + } +} + +/* + * HDA PCM callbacks + */ +static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct hdmi_spec *spec = codec->spec; + struct snd_pcm_runtime *runtime = substream->runtime; + int pin_idx, cvt_idx, mux_idx = 0; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + struct hdmi_spec_per_cvt *per_cvt = NULL; + int err; + + /* Validate hinfo */ + pin_idx = hinfo_to_pin_index(spec, hinfo); + if (snd_BUG_ON(pin_idx < 0)) + return -EINVAL; + per_pin = get_pin(spec, pin_idx); + eld = &per_pin->sink_eld; + + err = hdmi_choose_cvt(codec, pin_idx, &cvt_idx, &mux_idx); + if (err < 0) + return err; + + per_cvt = get_cvt(spec, cvt_idx); /* Claim converter */ per_cvt->assigned = 1; hinfo->nid = per_cvt->cvt_nid; @@ -1158,6 +1214,11 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, snd_hda_codec_write(codec, per_pin->pin_nid, 0, AC_VERB_SET_CONNECT_SEL, mux_idx); + + /* configure unused pins to choose other converters */ + if (codec->vendor_id == 0x80862807) + haswell_config_cvts(codec, pin_idx, mux_idx); + snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); /* Initially set the converter's capabilities */ -- 1.7.9.5

4 4

[alsa-devel] [PATCH 0/4] Haswell Display audio routing bug fix
by Wang Xingchao 18 Jun '13

18 Jun '13

This patchset used to fix some display audio issues on Haswell platform. I tested this patch on Haswell ult board, C0 stepping, with eDP pannel, HDMI monitor, DP monitor. The fixed issues: 1) when only HDMI or DP monitor connected, hear sound on ALL three HDMI devices. 2) when HDMI + DP monitors connected, hear sound on HDMI and DP at the same time no matter you're playing audio on DP or HDMI device. 3) When DP + HDMI + eDP connected, no sound could be heard. 4) After bootup, play audio on HDMI device 7(use second pin), no sound. If play audio on device 3 at first(use first pin), it has sound again. There's quite different hardware changes for haswell display audio, compared with previous Ivybridge/Sandybridge. The common hd-a driver and hdmi driver need make improvement to support. Haswell chip supports Dp1.2 feature(see ref 1). This is very cool feture windows had supported. We get confirm linux gfx team has plan to support Dp1.2 in future too. We're ready to support dp1.2 feature in audio side, as the converter selection dependency is fixed in this patchset. Also we need some improvement in HDA driver side because Haswell added new Verb to support DP1.2. here’s a video show about DP1.2 feature http://www.club-3d.com/index.php/products/reader.en/product/mst-hub-1-3.html Please apply the patches based on Daniel's drm-intel-next-queued branch: http://cgit.freedesktop.org/~danvet/drm-intel/log/?h=drm-intel-next-queued last commit 80e83831a64b9a5d49e844691037b2d4be0f14f9 Please feel free to let me know the issues you meet during test. Wang Xingchao (4): ALSA: hda - Haswell converter power state D0 verify ALSA: hda - Return error when open empty hdmi device drm/i915: Add display audio routing APIs for ALSA ALSA: hda - Add display audio routing API for haswell drivers/gpu/drm/i915/i915_drv.h | 18 +++++ drivers/gpu/drm/i915/intel_ddi.c | 131 +++++++++++++++++++++++++++++++++-- drivers/gpu/drm/i915/intel_display.c | 7 +- drivers/gpu/drm/i915/intel_drv.h | 1 + include/drm/i915_powerwell.h | 5 ++ sound/pci/hda/hda_i915.c | 83 ++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 4 ++ sound/pci/hda/patch_hdmi.c | 37 ++++++++++ 8 files changed, 281 insertions(+), 5 deletions(-) -- 1.8.1.2

6 17

[alsa-devel] [PATCH] ALSA: hda - Haswell converter power state D0 verify
by Wang Xingchao 18 Jun '13

18 Jun '13

Haswell converters maybe in wrong power state before usage. i.e. only converter 0 is in D0, converter 1/2 are in D3. When pin choose converter 1/2, there's no audio output, this cause dependency when playing differnt stream on pins. AUD_PWRST ConvertorA_Widget_Power_State_Current D0 AUD_PWRST ConvertorA_Widget_Power_State_Requsted D0 AUD_PWRST ConvertorB_Widget_Power_State_Current D3 AUD_PWRST ConvertorB_Widget_Power_State_Requested D3 AUD_PWRST ConvC_Widget_PwrSt_Curr D3 AUD_PWRST ConvC_Widget_PwrSt_Req D3 This patch check converter's power state and set D0 if it's in D3 mode. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/patch_hdmi.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a0..760eeb1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1018,10 +1018,19 @@ static void hdmi_unsol_event(struct hda_codec *codec, unsigned int res) hdmi_non_intrinsic_event(codec, res); } -static void haswell_verify_pin_D0(struct hda_codec *codec, hda_nid_t nid) +static void haswell_verify_pin_D0(struct hda_codec *codec, + hda_nid_t cvt_nid, hda_nid_t nid) { int pwr, lamp, ramp; + /* For Haswell, the converter 1/2 may keep in D3 state after bootup, + * thus pins could only choose converter 0 for use. Make sure the + * converters are in correct power state */ + pwr = snd_hda_codec_read(codec, cvt_nid, 0, AC_VERB_GET_POWER_STATE, 0); + pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; + if (pwr != AC_PWRST_D0) + snd_hda_codec_write(codec, cvt_nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + pwr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_POWER_STATE, 0); pwr = (pwr & AC_PWRST_ACTUAL) >> AC_PWRST_ACTUAL_SHIFT; if (pwr != AC_PWRST_D0) { @@ -1068,7 +1077,7 @@ static int hdmi_setup_stream(struct hda_codec *codec, hda_nid_t cvt_nid, int new_pinctl = 0; if (codec->vendor_id == 0x80862807) - haswell_verify_pin_D0(codec, pin_nid); + haswell_verify_pin_D0(codec, cvt_nid, pin_nid); if (snd_hda_query_pin_caps(codec, pin_nid) & AC_PINCAP_HBR) { pinctl = snd_hda_codec_read(codec, pin_nid, 0, -- 1.8.1.2

2 1

[alsa-devel] [PATCH v10 0/2] ARM: davinci: move EDMA API to arm/common
by Sekhar Nori 18 Jun '13

18 Jun '13

This is a repost of first two patches of Matt Porter's EDMA series below: http://www.mail-archive.com/davinci-linux-open-source@linux.davincidsp.com/… The only change that has been made is to drop changes related to DaVinci SFFSDR machine support for sound since that code is now queued to be removed. Thanks, Sekhar Matt Porter (2): ARM: davinci: move private EDMA API to arm/common ARM: edma: remove unused transfer controller handlers arch/arm/Kconfig | 1 + arch/arm/common/Kconfig | 3 + arch/arm/common/Makefile | 1 + arch/arm/{mach-davinci/dma.c => common/edma.c} | 39 +-------- arch/arm/mach-davinci/Makefile | 2 +- arch/arm/mach-davinci/board-tnetv107x-evm.c | 2 +- arch/arm/mach-davinci/davinci.h | 2 +- arch/arm/mach-davinci/devices-tnetv107x.c | 2 +- arch/arm/mach-davinci/devices.c | 6 +- arch/arm/mach-davinci/dm355.c | 2 +- arch/arm/mach-davinci/dm365.c | 2 +- arch/arm/mach-davinci/dm644x.c | 2 +- arch/arm/mach-davinci/dm646x.c | 2 +- arch/arm/mach-davinci/include/mach/da8xx.h | 2 +- drivers/dma/edma.c | 2 +- drivers/mmc/host/davinci_mmc.c | 1 + include/linux/mfd/davinci_voicecodec.h | 3 +- .../mach => include/linux/platform_data}/edma.h | 89 +------------------- include/linux/platform_data/spi-davinci.h | 2 +- sound/soc/davinci/davinci-evm.c | 1 + sound/soc/davinci/davinci-pcm.c | 1 + sound/soc/davinci/davinci-pcm.h | 2 +- 22 files changed, 29 insertions(+), 140 deletions(-) rename arch/arm/{mach-davinci/dma.c => common/edma.c} (97%) rename {arch/arm/mach-davinci/include/mach => include/linux/platform_data}/edma.h (59%) -- 1.7.10.1

4 7

[alsa-devel] [PATCH 1/2] ASoC: spear: Remove nonexistant EVM options
by Mark Brown 18 Jun '13

18 Jun '13

The source wasn't added. Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/spear/Kconfig | 21 --------------------- sound/soc/spear/Makefile | 3 --- 2 files changed, 24 deletions(-) diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index e3851d6..3567d73 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -1,24 +1,3 @@ -config SND_SPEAR_EVM - tristate "SoC Audio support for SPEAr EVM" - select SND_DESIGNWARE_I2S - select SND_SOC_STA529 - select SND_SPEAR_SOC - help - Say Y if you want to add support for SoC audio on SPEAr - platform - -config SND_SPEAR1340_EVM - tristate "SoC Audio support for SPEAr1340 EVM" - select SND_DESIGNWARE_I2S - select SND_SOC_STA529 - select SND_SPEAR_SPDIF_OUT - select SND_SPEAR_SPDIF_IN - select SND_SOC_SPDIF - select SND_SPEAR_SOC - help - Say Y if you want to add support for SoC audio on SPEAr1340 - platform - config SND_SPEAR_SOC tristate select SND_SOC_DMAENGINE_PCM diff --git a/sound/soc/spear/Makefile b/sound/soc/spear/Makefile index b365126..9f47bb4 100644 --- a/sound/soc/spear/Makefile +++ b/sound/soc/spear/Makefile @@ -2,6 +2,3 @@ obj-$(CONFIG_SND_SPEAR_SOC) += spear_pcm.o obj-$(CONFIG_SND_SPEAR_SPDIF_IN) += spdif_in.o obj-$(CONFIG_SND_SPEAR_SPDIF_OUT) += spdif_out.o - -# SPEAR Machine Support -obj-$(CONFIG_SND_SPEAR_EVM) += spear_evb.o -- 1.7.10.4

2 3

[alsa-devel] VIA VT1708S mic booast problem
by kbd_server＠rbcmail.ru 18 Jun '13

18 Jun '13

Dears, Please, check bug report - https://bugzilla.kernel.org/show_bug.cgi?id=59861 In 3.9.5-030905-generic version - Rear (Front) Mic Boost Volume option is absent for VIA VT1708S codec. 3.8.0-19-generic - everything OK. -- Почтовый ящик предоставлен сервисом qip.ru Заведи бесплатную почту в домене rbcmail.ru и ты!

1 0

Re: [alsa-devel] [PATCH 1/3 v2] mfd: arizona: Integrate wm8997 into Arizona mfd
by Samuel Ortiz 18 Jun '13

18 Jun '13

Hi Charles, On Thu, Jun 13, 2013 at 09:43:29AM +0100, Charles Keepax wrote: > The wm8997 is a compact, high-performance audio hub CODEC with SLIMbus > interfacing, for smartphones, tablets and other portable audio devices > based on the Arizona platform. > > This patch integrates the wm8997 into the Arizona mfd. > > Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> > --- > drivers/mfd/Kconfig | 6 + > drivers/mfd/Makefile | 3 + > drivers/mfd/arizona-core.c | 27 + > drivers/mfd/arizona-i2c.c | 6 + > drivers/mfd/arizona-irq.c | 8 + > drivers/mfd/arizona.h | 5 + > drivers/mfd/wm8997-tables.c | 1525 ++++++++++++++++++++++++++++++++++++++ > include/linux/mfd/arizona/core.h | 2 + > 8 files changed, 1582 insertions(+), 0 deletions(-) > create mode 100644 drivers/mfd/wm8997-tables.c Applied to mfd-next, thanks. Cheers, Samuel. -- Intel Open Source Technology Centre http://oss.intel.com/

1 0

[alsa-devel] [PATCH 1/2] mfd: wm5102: Expose DRE control registers
by Charles Keepax 18 Jun '13

18 Jun '13

Certain use cases may require specific DRE settings so expose the necessary registers. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm5102-tables.c | 9 ++++++- include/linux/mfd/arizona/registers.h | 44 +++++++++++++++++++++++++++++++++ 2 files changed, 52 insertions(+), 1 deletions(-) diff --git a/drivers/mfd/wm5102-tables.c b/drivers/mfd/wm5102-tables.c index 155c4a1..802dd3c 100644 --- a/drivers/mfd/wm5102-tables.c +++ b/drivers/mfd/wm5102-tables.c @@ -65,7 +65,8 @@ static const struct reg_default wm5102_revb_patch[] = { { 0x418, 0xa080 }, { 0x420, 0xa080 }, { 0x428, 0xe000 }, - { 0x443, 0xDC1A }, + { 0x442, 0x3F0A }, + { 0x443, 0xDC1F }, { 0x4B0, 0x0066 }, { 0x458, 0x000b }, { 0x212, 0x0000 }, @@ -424,6 +425,9 @@ static const struct reg_default wm5102_reg_default[] = { { 0x00000435, 0x0180 }, /* R1077 - DAC Digital Volume 5R */ { 0x00000436, 0x0081 }, /* R1078 - DAC Volume Limit 5R */ { 0x00000437, 0x0200 }, /* R1079 - Noise Gate Select 5R */ + { 0x00000440, 0x8FFF }, /* R1088 - DRE Enable */ + { 0x00000442, 0x3F0A }, /* R1090 - DRE Control 2 */ + { 0x00000443, 0xDC1F }, /* R1090 - DRE Control 3 */ { 0x00000450, 0x0000 }, /* R1104 - DAC AEC Control 1 */ { 0x00000458, 0x000B }, /* R1112 - Noise Gate Control */ { 0x00000490, 0x0069 }, /* R1168 - PDM SPK1 CTRL 1 */ @@ -1197,6 +1201,9 @@ static bool wm5102_readable_register(struct device *dev, unsigned int reg) case ARIZONA_DAC_DIGITAL_VOLUME_5R: case ARIZONA_DAC_VOLUME_LIMIT_5R: case ARIZONA_NOISE_GATE_SELECT_5R: + case ARIZONA_DRE_ENABLE: + case ARIZONA_DRE_CONTROL_2: + case ARIZONA_DRE_CONTROL_3: case ARIZONA_DAC_AEC_CONTROL_1: case ARIZONA_NOISE_GATE_CONTROL: case ARIZONA_PDM_SPK1_CTRL_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 4730b5c..4706d3d 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -215,6 +215,9 @@ #define ARIZONA_DAC_DIGITAL_VOLUME_6R 0x43D #define ARIZONA_DAC_VOLUME_LIMIT_6R 0x43E #define ARIZONA_NOISE_GATE_SELECT_6R 0x43F +#define ARIZONA_DRE_ENABLE 0x440 +#define ARIZONA_DRE_CONTROL_2 0x442 +#define ARIZONA_DRE_CONTROL_3 0x443 #define ARIZONA_DAC_AEC_CONTROL_1 0x450 #define ARIZONA_NOISE_GATE_CONTROL 0x458 #define ARIZONA_PDM_SPK1_CTRL_1 0x490 @@ -3133,6 +3136,47 @@ #define ARIZONA_OUT6R_NGATE_SRC_WIDTH 12 /* OUT6R_NGATE_SRC - [11:0] */ /* + * R1088 (0x440) - DRE Enable + */ +#define ARIZONA_DRE3L_ENA 0x0010 /* DRE3L_ENA */ +#define ARIZONA_DRE3L_ENA_MASK 0x0010 /* DRE3L_ENA */ +#define ARIZONA_DRE3L_ENA_SHIFT 4 /* DRE3L_ENA */ +#define ARIZONA_DRE3L_ENA_WIDTH 1 /* DRE3L_ENA */ +#define ARIZONA_DRE2R_ENA 0x0008 /* DRE2R_ENA */ +#define ARIZONA_DRE2R_ENA_MASK 0x0008 /* DRE2R_ENA */ +#define ARIZONA_DRE2R_ENA_SHIFT 3 /* DRE2R_ENA */ +#define ARIZONA_DRE2R_ENA_WIDTH 1 /* DRE2R_ENA */ +#define ARIZONA_DRE2L_ENA 0x0004 /* DRE2L_ENA */ +#define ARIZONA_DRE2L_ENA_MASK 0x0004 /* DRE2L_ENA */ +#define ARIZONA_DRE2L_ENA_SHIFT 2 /* DRE2L_ENA */ +#define ARIZONA_DRE2L_ENA_WIDTH 1 /* DRE2L_ENA */ +#define ARIZONA_DRE1R_ENA 0x0002 /* DRE1R_ENA */ +#define ARIZONA_DRE1R_ENA_MASK 0x0002 /* DRE1R_ENA */ +#define ARIZONA_DRE1R_ENA_SHIFT 1 /* DRE1R_ENA */ +#define ARIZONA_DRE1R_ENA_WIDTH 1 /* DRE1R_ENA */ +#define ARIZONA_DRE1L_ENA 0x0001 /* DRE1L_ENA */ +#define ARIZONA_DRE1L_ENA_MASK 0x0001 /* DRE1L_ENA */ +#define ARIZONA_DRE1L_ENA_SHIFT 0 /* DRE1L_ENA */ +#define ARIZONA_DRE1L_ENA_WIDTH 1 /* DRE1L_ENA */ + +/* + * R1090 (0x442) - DRE Control 2 + */ +#define ARIZONA_DRE_T_LOW_MASK 0x3F00 /* DRE_T_LOW - [13:8] */ +#define ARIZONA_DRE_T_LOW_SHIFT 8 /* DRE_T_LOW - [13:8] */ +#define ARIZONA_DRE_T_LOW_WIDTH 6 /* DRE_T_LOW - [13:8] */ + +/* + * R1091 (0x443) - DRE Control 3 + */ +#define ARIZONA_DRE_GAIN_SHIFT_MASK 0xC000 /* DRE_GAIN_SHIFT - [15:14] */ +#define ARIZONA_DRE_GAIN_SHIFT_SHIFT 14 /* DRE_GAIN_SHIFT - [15:14] */ +#define ARIZONA_DRE_GAIN_SHIFT_WIDTH 2 /* DRE_GAIN_SHIFT - [15:14] */ +#define ARIZONA_DRE_LOW_LEVEL_ABS_MASK 0x000F /* LOW_LEVEL_ABS - [3:0] */ +#define ARIZONA_DRE_LOW_LEVEL_ABS_SHIFT 0 /* LOW_LEVEL_ABS - [3:0] */ +#define ARIZONA_DRE_LOW_LEVEL_ABS_WIDTH 4 /* LOW_LEVEL_ABS - [3:0] */ + +/* * R1104 (0x450) - DAC AEC Control 1 */ #define ARIZONA_AEC_LOOPBACK_SRC_MASK 0x003C /* AEC_LOOPBACK_SRC - [5:2] */ -- 1.7.2.5

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