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

[alsa-devel] [PATCH] ALSA: emu10k1: handle dock disconnects
by Michael Gernoth 11 Apr '15

11 Apr '15

When the dock on an E-mu 1010 card is disconnected, all outputs get muted by the hardware. Add logic to detect a disconnect and unmute. Signed-off-by: Michael Gernoth <michael(a)gernoth.net> --- sound/pci/emu10k1/emu10k1_main.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index b4458a6..ee50c1d 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -707,6 +707,7 @@ static int emu1010_firmware_thread(void *data) { struct snd_emu10k1 *emu = data; u32 tmp, tmp2, reg; + u32 last_reg = 0; int err; for (;;) { @@ -782,7 +783,15 @@ static int emu1010_firmware_thread(void *data) msleep(10); /* Unmute all. Default is muted after a firmware load */ snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); + } else if (!reg && last_reg) { + /* Audio Dock removed */ + dev_info(emu->card->dev, + "emu1010: Audio Dock detached\n"); + /* Unmute all */ + snd_emu1010_fpga_write(emu, EMU_HANA_UNMUTE, EMU_UNMUTE); } + + last_reg = reg; } dev_info(emu->card->dev, "emu1010: firmware thread stopping\n"); return 0; -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: seq: fill client ID in return value of pool operation
by Takashi Sakamoto 11 Apr '15

11 Apr '15

The returned value of 'get/seq client pool' operation has zeroed value for its client ID, against requested client ID. This commit fix the bug by filling it with index value of referred client object. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/seq/seq_clientmgr.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 4828765..edbdab8 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1879,6 +1879,7 @@ static int snd_seq_ioctl_get_client_pool(struct snd_seq_client *client, if (cptr == NULL) return -ENOENT; memset(&info, 0, sizeof(info)); + info.client = cptr->number; info.output_pool = cptr->pool->size; info.output_room = cptr->pool->room; info.output_free = info.output_pool; -- 2.1.0

2 1

[alsa-devel] [PATCH 0/2] Cygnus Audio Driver
by Scott Branden 11 Apr '15

11 Apr '15

Please provide comments on the initial version of this driver. This patchset contains the devicetree bindings and core audio driver for the Cygnus SoC. There is an open question on how to fit this driver into the clock framework (if at all). The audio PLL is embedded in the audio block and only used by the audio block. The audio PLL registers are also in the middle of the audio register map. In addition, the audio PLL is adjustable to less than 1 Hz. The existing clock driver framework does not provide a mechanism to take advantage of the resolution of the hardware. Can the audio PLL remain within the audio driver and/or any modifications required? Lori Hikichi (2): ASoC: cygnus-audio: adding device tree bindings ASoC: add core audio driver for Broadcom Cygnus SOC. .../bindings/sound/brcm,cygnus-audio.txt | 68 + sound/soc/bcm/Kconfig | 11 + sound/soc/bcm/Makefile | 5 +- sound/soc/bcm/cygnus-pcm.c | 918 +++++++++++ sound/soc/bcm/cygnus-pcm.h | 45 + sound/soc/bcm/cygnus-ssp.c | 1613 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 84 + 7 files changed, 2743 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-pcm.h create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 2.3.3

4 10

Re: [alsa-devel] [PATCH v3] ASoC: max98090: add shutdown callback for max98090
by anish kumar 11 Apr '15

11 Apr '15

On Fri, Apr 10, 2015 at 5:45 PM, anish singh <anish198519851985(a)gmail.com> wrote: > > ---------- Forwarded message ---------- > From: Caesar Wang <wxt(a)rock-chips.com> > Date: Wed, Apr 8, 2015 at 4:05 AM > Subject: [PATCH v3] ASoC: max98090: add shutdown callback for max98090 > To: Sonny Rao <sonnyrao(a)chromium.org>, Cheng-Yi Chiang > <cychiang(a)chromium.org>, Heiko Stuebner <heiko(a)sntech.de>, Mark Brown > <broonie(a)kernel.org>, zhengxing <zhengxing(a)rock-chips.com>, "jay.xu" > <xjq(a)rock-chips.com> > Cc: devicetree(a)vger.kernel.org, dianders(a)chromium.org, > dri-devel(a)lists.freedesktop.org, linux-kernel(a)vger.kernel.org, > linux-rockchip(a)lists.infradead.org, linux-arm-kernel(a)lists.infradead.org, > Caesar Wang <wxt(a)rock-chips.com>, Jaroslav Kysela <perex(a)perex.cz>, > alsa-devel(a)alsa-project.org, Takashi Iwai <tiwai(a)suse.de>, Dylan Reid > <dgreid(a)chromium.org>, Liam Girdwood <lgirdwood(a)gmail.com>, Lars-Peter > Clausen <lars(a)metafoo.de>, Jarkko Nikula <jarkko.nikula(a)linux.intel.com>, > Tushar Behera <tushar.b(a)samsung.com> > > > To fix pop noise when shutdown,the pop noise during shutdown > is the pmic cutoff power of codec without any notice. > > Signed-off-by: jay.xu <xjq(a)rock-chips.com> > Signed-off-by: zhengxing <zhengxing(a)rock-chips.com> > Signed-off-by: Caesar Wang <wxt(a)rock-chips.com> > > --- > > Changes in v3: > - modify the shutdown function before remove(..) > - fix the `Serien-cc` > > Changes in v2: > - remove the dev_info(..) > - fix the comment style > - add the max98090_i2c_shutdown() in remove() fuction > > sound/soc/codecs/max98090.c | 17 +++++++++++++++++ > 1 file changed, 17 insertions(+) > > diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c > index b112b1c..3e33ef2 100644 > --- a/sound/soc/codecs/max98090.c > +++ b/sound/soc/codecs/max98090.c > @@ -2605,8 +2605,24 @@ err_enable: > return ret; > } > Please check attached patch and see if it solves your problem. We tested it here with zcd on. +static void max98090_i2c_shutdown(struct i2c_client *i2c) +{ + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); + + dev_info(&i2c->dev, "shut down device\n"); + + /* enable zero crossing */ + regmap_write(max98090->regmap, + M98090_REG_LEVEL_CONTROL, M98090_ZDENN_MASK-1); + + /* turn down volume levels for all the outputs */ + regmap_write(max98090->regmap, + M98090_REG_RCV_LOUTL_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_LOUTR_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_LEFT_SPK_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_RIGHT_SPK_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_LEFT_HP_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_RIGHT_HP_VOLUME, 0x00); + regmap_write(max98090->regmap, + M98090_REG_DEVICE_SHUTDOWN, 0x00); + msleep(40); +} + if it doesn't solve your problem then please dump your registers and send it to us to look further. I assume dapm is turning off the widgets before shutting down the codec. > +static void max98090_i2c_shutdown(struct i2c_client *i2c) > +{ > + struct max98090_priv *max98090 = dev_get_drvdata(&i2c->dev); > + > + /* > + * Enable volume smoothing, disable zero cross. This will cause > + * a quick 40ms ramp to mute on shutdown. > + */ > + regmap_write(max98090->regmap, > + M98090_REG_LEVEL_CONTROL, M98090_VSENN_MASK); > + regmap_write(max98090->regmap, > + M98090_REG_DEVICE_SHUTDOWN, 0x00); > + msleep(40); > +} > + > static int max98090_i2c_remove(struct i2c_client *client) > { > + max98090_i2c_shutdown(client); > snd_soc_unregister_codec(&client->dev); > return 0; > } > @@ -2696,6 +2712,7 @@ static struct i2c_driver max98090_i2c_driver = { > .acpi_match_table = ACPI_PTR(max98090_acpi_match), > }, > .probe = max98090_i2c_probe, > + .shutdown = max98090_i2c_shutdown, > .remove = max98090_i2c_remove, > .id_table = max98090_i2c_id, > }; > -- > 1.9.1 > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ >

1 0

[alsa-devel] [PATCH] ASoC: codecs-ac97: Remove rate constraints
by Maciej S. Szmigiero 10 Apr '15

10 Apr '15

Remove rate constraints from generic ASoC AC'97 CODEC and make it selectable in config. Supported rates should be detected and constrained anyway by AC'97 generic code - was tested with VT1613 CODEC and iMX6 SSI controller. This way this driver can be used for platforms which don't need specialized AC'97 CODEC drivers while at the same avoiding code duplication from implementing equivalent functionality in a controller driver. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index d0ac723..5b3224c 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -44,10 +44,6 @@ static int ac97_prepare(struct snd_pcm_substream *substream, return snd_ac97_set_rate(ac97, reg, substream->runtime->rate); } -#define STD_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000) - static const struct snd_soc_dai_ops ac97_dai_ops = { .prepare = ac97_prepare, }; @@ -58,13 +54,13 @@ static struct snd_soc_dai_driver ac97_dai = { .stream_name = "AC97 Playback", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .capture = { .stream_name = "AC97 Capture", .channels_min = 1, .channels_max = 2, - .rates = STD_AC97_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SND_SOC_STD_AC97_FMTS,}, .ops = &ac97_dai_ops, }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 061c465..84cad9a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 select SND_SOC_AB8500_CODEC if ABX500_CORE - select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS + select SND_SOC_AC97_CODEC select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X_SPI if SPI_MASTER select SND_SOC_AD193X_I2C if I2C @@ -211,8 +211,9 @@ config SND_SOC_AB8500_CODEC tristate config SND_SOC_AC97_CODEC - tristate + tristate "Build generic ASoC AC97 CODEC driver" select SND_AC97_CODEC + select SND_SOC_AC97_BUS config SND_SOC_AD1836 tristate

1 0

[alsa-devel] [PATCH 0/3] ASoC: Intel: IPC/mailbox code refactoring
by Jin Yao 10 Apr '15

10 Apr '15

Hi Mark, Currently in Intel SST driver, some very similar or duplicated IPC/mailbox processing code are used in different platforms. This is a patch series which refactors the common IPC/mailbox code into generic APIs and use the new APIs in Baytrail and Broadwell platforms. This patch series depends on Jie Yang's following patches for build: ASoC: Intel: create atom folder and move atom platform files in ASoC: Intel: create baytrail folder and move baytrail platform files in ASoC: Intel: create boards folder and move sst boards files in ASoC: Intel: create haswell folder and move haswell platform files in ASoC: Intel: create common folder and move common files in Since now Jie Yang's patch series has been applied in intel branch so this patch series will be no build issue. Thanks Jin Yao Jin Yao (3): ASoC: Intel: Refactor common IPC/mailbox code into generic APIs ASoC: Intel: Use the generic IPC/mailbox APIs in Baytrail ASoC: Intel: Use the generic IPC/mailbox APIs in Broadwell sound/soc/intel/baytrail/sst-baytrail-ipc.c | 360 ++++++-------------------- sound/soc/intel/common/Makefile | 3 +- sound/soc/intel/common/sst-ipc.c | 294 +++++++++++++++++++++ sound/soc/intel/common/sst-ipc.h | 91 +++++++ sound/soc/intel/haswell/sst-haswell-ipc.c | 382 +++++++--------------------- 5 files changed, 551 insertions(+), 579 deletions(-) create mode 100644 sound/soc/intel/common/sst-ipc.c create mode 100644 sound/soc/intel/common/sst-ipc.h -- 1.9.1

4 7

[alsa-devel] [PATCH] ASoC: fsl: Add the audio interfacce acronyms in Kconfig text
by Fabio Estevam 10 Apr '15

10 Apr '15

To keep consistency with the other Kconfig entries, use the audio interfacce acronyms (SSI and SPDIF) in the Kconfig menu text. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 081e406..fa44fc6 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -24,7 +24,7 @@ config SND_SOC_FSL_SAI in-tree drivers select it automatically. config SND_SOC_FSL_SSI - tristate "Synchronous Serial Interface module support" + tristate "Synchronous Serial Interface module (SSI) support" select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) select REGMAP_MMIO @@ -35,7 +35,7 @@ config SND_SOC_FSL_SSI in-tree drivers select it automatically. config SND_SOC_FSL_SPDIF - tristate "Sony/Philips Digital Interface module support" + tristate "Sony/Philips Digital Interface (SPDIF) module support" select REGMAP_MMIO select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC) -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: soc-core: call snd_soc_runtime_set_dai_fmt() before soc_new_pcm()
by Kuninori Morimoto 10 Apr '15

10 Apr '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current snd_soc_runtime_set_dai_fmt() is called after soc_probe_link_dais(). this means snd_soc_dai_set_fmt() will be called after soc_new_pcm(). Before appling 1efb53a220b78fdfdbb97b726a2156713e75bdab (ASoC: simple-card: Remove support for setting differing DAI formats) simple-card user had (1) snd_soc_dai_set_fmt() -> soc_new_pcm(), but, after that it is (2) soc_new_pcm() -> snd_soc_dai_set_fmt(). At least rsnd driver is assuming (1) pattern. This patch move snd_soc_dai_set_fmt() into soc_probe_link_dais() after the dai_link->init section to solve this issue. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- >> Mark This patch solves my issue, but I don't know how much effect it has to other drivers. sound/soc/soc-core.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 76bfff2..9777e78 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1324,6 +1324,9 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) } } + if (dai_link->dai_fmt) + snd_soc_runtime_set_dai_fmt(rtd, dai_link->dai_fmt); + ret = soc_post_component_init(rtd, dai_link->name); if (ret) return ret; @@ -1642,12 +1645,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, card->num_of_dapm_routes); - for (i = 0; i < card->num_links; i++) { - if (card->dai_link[i].dai_fmt) - snd_soc_runtime_set_dai_fmt(&card->rtd[i], - card->dai_link[i].dai_fmt); - } - snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl_ssi: Use devm_ioremap_resource()
by Fabio Estevam 10 Apr '15

10 Apr '15

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using platform_get_resource() and devm_ioremap_resource() can make the code a bit simpler. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 20 ++++++-------------- 1 file changed, 6 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 143cc16..e8bb8ee 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1288,7 +1288,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) const struct of_device_id *of_id; const char *p, *sprop; const uint32_t *iprop; - struct resource res; + struct resource *res; void __iomem *iomem; char name[64]; @@ -1335,19 +1335,11 @@ static int fsl_ssi_probe(struct platform_device *pdev) } ssi_private->cpu_dai_drv.name = dev_name(&pdev->dev); - /* Get the addresses and IRQ */ - ret = of_address_to_resource(np, 0, &res); - if (ret) { - dev_err(&pdev->dev, "could not determine device resources\n"); - return ret; - } - ssi_private->ssi_phys = res.start; - - iomem = devm_ioremap(&pdev->dev, res.start, resource_size(&res)); - if (!iomem) { - dev_err(&pdev->dev, "could not map device resources\n"); - return -ENOMEM; - } + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + iomem = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(iomem)) + return PTR_ERR(iomem); + ssi_private->ssi_phys = res->start; ret = of_property_match_string(np, "clock-names", "ipg"); if (ret < 0) { -- 1.9.1

2 1

Re: [alsa-devel] [Intel-gfx] [PATCH 15/19] drm/i915: HSW cdclk support
by Takashi Iwai 10 Apr '15

10 Apr '15

At Fri, 10 Apr 2015 16:27:39 +0300, Mika Kahola wrote: > > On Thu, 2015-04-09 at 17:17 +0200, Takashi Iwai wrote: > > At Thu, 9 Apr 2015 15:51:27 +0200, > > Daniel Vetter wrote: > > > > > > On Thu, Apr 09, 2015 at 04:41:26PM +0300, Mika Kahola wrote: > > > > On Thu, 2015-04-09 at 11:32 +0200, Daniel Vetter wrote: > > > > > On Thu, Apr 09, 2015 at 10:24:24AM +0300, Mika Kahola wrote: > > > > > > I did some testing on audio part with HDMI-HDMI and DP-HDMI cables > > > > > > connected to my Haswell box. Before applying the patch I tested as a > > > > > > reference with the latest -nightly (04-08-2015), 4.0-rc7. Unfortunately, > > > > > > I failed to get any audio over HDMI cable. For a reference I tested with > > > > > > the very same setup the vanillla kernel from Linus tree 4.0-rc7 and with > > > > > > that kernel the audio worked ok. Then I did some GIT bisecting and it > > > > > > turned out that the first commit that I failed to get audio working was > > > > > > aa2fee4286e43b4784982b17669b02cc99c1ae55. > > > > > > > > > I rerun the bisecting and this time the result was > > > > > > > > commit 0a599838737a2527c35e4d94f794aefe59df1781 > > > > Merge: 2d846c7 a59d719 > > > > Author: Takashi Iwai <tiwai(a)suse.de> > > > > Date: Wed Apr 8 11:29:56 2015 +0200 > > > > > > > > Merge branch 'for-linus' into for-next > > > > > > > > Back merge HD-audio quirks to for-next branch, so that we can apply > > > > a couple of more quirks. > > > > > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > > > > > Adding Takashi and intel audio folks. > > > > The bisecting looks odd. The commit you pointed is a back-merge from > > 4.0-rc to next branch, so this merge itself shouldn't bring so many > > stuff -- at least about the sound part. > > > > The diff in sound/* is found below. As you can see, the only change > > relevant with HDMI is the chunk in sound/pci/hda/hda_intel.c for > > HD-audio controller, but it's specific to Skylake, thus this must be > > irrelevant with your hardware. > > > > Please double-check. > > I double-checked the bisecting and I ended up with the same result. You're > right, it does seems odd why audio on my Haswell box stopped working. > > Any ideas how to proceed from here? Try to check whether the reverting the diff of sound/* manually fixes the problem. (i.e. apply diff -R for the diff I showed in the previous mail.) If it doesn't fix but still reverting the whole commit does fix, it means that something else got broken by the merge. Takashi > > Cheers, > Mika > > > > thanks, > > > > Takashi > > > > === > > % git diff 0a599838737a2527c35e4d94f794aefe59df1781^..0a599838737a2527c35e4d94f794aefe59df1781 sound > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > > index 9bcc5457a83e..e1c210515581 100644 > > --- a/sound/pci/hda/hda_intel.c > > +++ b/sound/pci/hda/hda_intel.c > > @@ -1967,7 +1967,7 @@ static const struct pci_device_id azx_ids[] = { > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > /* Sunrise Point */ > > { PCI_DEVICE(0x8086, 0xa170), > > - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, > > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, > > /* Sunrise Point-LP */ > > { PCI_DEVICE(0x8086, 0x9d70), > > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > > index e0c06f9a0e80..7f46d063af57 100644 > > --- a/sound/pci/hda/patch_realtek.c > > +++ b/sound/pci/hda/patch_realtek.c > > @@ -396,7 +396,7 @@ static void alc_auto_setup_eapd(struct hda_codec *codec, bool on) > > { > > /* We currently only handle front, HP */ > > static hda_nid_t pins[] = { > > - 0x0f, 0x10, 0x14, 0x15, 0 > > + 0x0f, 0x10, 0x14, 0x15, 0x17, 0 > > }; > > hda_nid_t *p; > > for (p = pins; *p; p++) > > @@ -2870,6 +2870,8 @@ static void alc283_init(struct hda_codec *codec) > > > > if (!hp_pin) > > return; > > + > > + msleep(30); > > hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); > > > > /* Index 0x43 Direct Drive HP AMP LPM Control 1 */ > > @@ -3564,6 +3566,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > break; > > case 0x10ec0233: > > @@ -3619,6 +3622,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_write_coef_idx(codec, 0x45, 0xc489); > > snd_hda_set_pin_ctl_cache(codec, hp_pin, 0); > > alc_process_coef_fw(codec, coef0255); > > @@ -3688,6 +3692,7 @@ static void alc_headset_mode_default(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > break; > > case 0x10ec0233: > > @@ -3742,6 +3747,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > break; > > case 0x10ec0233: > > @@ -3796,6 +3802,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > break; > > case 0x10ec0233: > > @@ -3841,6 +3848,7 @@ static void alc_determine_headset_type(struct hda_codec *codec) > > > > switch (codec->core.vendor_id) { > > case 0x10ec0255: > > + case 0x10ec0256: > > alc_process_coef_fw(codec, coef0255); > > msleep(300); > > val = alc_read_coef_idx(codec, 0x46); > > @@ -4993,6 +5001,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > > SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), > > SND_PCI_QUIRK(0x17aa, 0x501e, "Thinkpad L440", ALC292_FIXUP_TPT440_DOCK), > > SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > > + SND_PCI_QUIRK(0x17aa, 0x5036, "Thinkpad T450s", ALC292_FIXUP_TPT440_DOCK), > > SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), > > SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), > > SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), > > @@ -5173,6 +5182,16 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { > > {0x17, 0x40000000}, > > {0x1d, 0x40700001}, > > {0x21, 0x02211050}), > > + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, > > + {0x12, 0x90a60140}, > > + {0x13, 0x40000000}, > > + {0x14, 0x90170110}, > > + {0x19, 0x411111f0}, > > + {0x1a, 0x411111f0}, > > + {0x1b, 0x411111f0}, > > + {0x1d, 0x40700001}, > > + {0x1e, 0x411111f0}, > > + {0x21, 0x02211020}), > > SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, > > {0x12, 0x90a60130}, > > {0x13, 0x40000000}, > > diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c > > index 353532b8aee4..8f63e8c8b46c 100644 > > --- a/sound/usb/quirks.c > > +++ b/sound/usb/quirks.c > > @@ -1113,8 +1113,13 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, > > > > bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) > > { > > - /* MS Lifecam HD-5000 doesn't support reading the sample rate. */ > > - return chip->usb_id == USB_ID(0x045E, 0x076D); > > + /* devices which do not support reading the sample rate. */ > > + switch (chip->usb_id) { > > + case USB_ID(0x045E, 0x076D): /* MS Lifecam HD-5000 */ > > + case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ > > + return true; > > + } > > + return false; > > } > > > > /* Marantz/Denon USB DACs need a vendor cmd to switch > > > >

1 0