October 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 17 Oct '19

17 Oct '19

For P2P output, the output divider should align with the output sample rate, if use ideal sample rate, there will be a lot of overload, which would cause underrun. The maximum divider of asrc clock is 1024, but there is no judgement for this limitaion in driver, which may cause the divider setting not correct. For non-ideal ratio mode, the clock rate should divide the sample rate with no remainder, and the quotient should be less than 1024. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_asrc.c | 40 +++++++++++++++++++++++++++++++--------- 1 file changed, 31 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index 0bf91a6f54b9..44d05ec28bd3 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -260,7 +260,7 @@ static int fsl_asrc_set_ideal_ratio(struct fsl_asrc_pair *pair, * of struct asrc_config which includes in/output sample rate, width, channel * and clock settings. */ -static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) +static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair, bool p2p) { struct asrc_config *config = pair->config; struct fsl_asrc *asrc_priv = pair->asrc_priv; @@ -268,7 +268,8 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) enum asrc_word_width input_word_width; enum asrc_word_width output_word_width; u32 inrate, outrate, indiv, outdiv; - u32 clk_index[2], div[2]; + u32 clk_index[2], div[2], rem[2]; + u64 clk_rate; int in, out, channels; int pre_proc, post_proc; struct clk *clk; @@ -351,7 +352,9 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) /* We only have output clock for ideal ratio mode */ clk = asrc_priv->asrck_clk[clk_index[ideal ? OUT : IN]]; - div[IN] = clk_get_rate(clk) / inrate; + clk_rate = clk_get_rate(clk); + rem[IN] = do_div(clk_rate, inrate); + div[IN] = (u32)clk_rate; if (div[IN] == 0) { pair_err("failed to support input sample rate %dHz by asrck_%x\n", inrate, clk_index[ideal ? OUT : IN]); @@ -360,11 +363,20 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) clk = asrc_priv->asrck_clk[clk_index[OUT]]; - /* Use fixed output rate for Ideal Ratio mode (INCLK_NONE) */ - if (ideal) - div[OUT] = clk_get_rate(clk) / IDEAL_RATIO_RATE; - else - div[OUT] = clk_get_rate(clk) / outrate; + /* + * When P2P mode, output rate should align with the out samplerate. + * if set too high output rate, there will be lots of Overload. + * When M2M mode, output rate should also need to align with the out + * samplerate, but M2M must use less time to achieve good performance. + */ + clk_rate = clk_get_rate(clk); + if (p2p || !ideal) { + rem[OUT] = do_div(clk_rate, outrate); + div[OUT] = clk_rate; + } else { + rem[OUT] = do_div(clk_rate, IDEAL_RATIO_RATE); + div[OUT] = clk_rate; + } if (div[OUT] == 0) { pair_err("failed to support output sample rate %dHz by asrck_%x\n", @@ -372,6 +384,16 @@ static int fsl_asrc_config_pair(struct fsl_asrc_pair *pair) return -EINVAL; } + if (!ideal && (div[IN] > 1024 || div[OUT] > 1024 || + rem[IN] != 0 || rem[OUT] != 0)) { + pair_err("The divider can't be used for non ideal mode\n"); + return -EINVAL; + } + + /* Divider range is [1, 1024] */ + div[IN] = min_t(u32, 1024, div[IN]); + div[OUT] = min_t(u32, 1024, div[OUT]); + /* Set the channel number */ channels = config->channel_num; @@ -560,7 +582,7 @@ static int fsl_asrc_dai_hw_params(struct snd_pcm_substream *substream, config.output_sample_rate = rate; } - ret = fsl_asrc_config_pair(pair); + ret = fsl_asrc_config_pair(pair, true); if (ret) { dev_err(dai->dev, "fail to config asrc pair\n"); return ret; -- 2.21.0

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[alsa-devel] [PATCH v3] dt-bindings: sound: Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to dt-schema
by Marek Szyprowski 17 Oct '19

17 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> [mszyprow: reordered non-standard properties, added list of values for widgets, minor other fixes] Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,tm2-audio.txt | 42 -------- .../bindings/sound/samsung,tm2-audio.yaml | 99 +++++++++++++++++++ 2 files changed, 99 insertions(+), 42 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt deleted file mode 100644 index f5ccc12ddc00..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt +++ /dev/null @@ -1,42 +0,0 @@ -Samsung Exynos5433 TM2(E) audio complex with WM5110 codec - -Required properties: - - - compatible : "samsung,tm2-audio" - - model : the user-visible name of this sound complex - - audio-codec : the first entry should be phandle of the wm5110 audio - codec node, as described in ../mfd/arizona.txt; - the second entry should be phandle of the HDMI - transmitter node - - i2s-controller : the list of phandle and argument tuples pointing to - I2S controllers, the first entry should be I2S0 and - the second one I2S1 - - audio-amplifier : the phandle of the MAX98504 amplifier - - samsung,audio-routing : a list of the connections between audio components; - each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source; valid names for sources and sinks are the - WM5110's and MAX98504's pins and the jacks on the - board: HP, SPK, Main Mic, Sub Mic, Third Mic, - Headset Mic - - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator - - -Example: - -sound { - compatible = "samsung,tm2-audio"; - audio-codec = <&wm5110>, <&hdmi>; - i2s-controller = <&i2s0 0>, <&i2s1 0>; - audio-amplifier = <&max98504>; - mic-bias-gpios = <&gpr3 2 0>; - model = "wm5110"; - samsung,audio-routing = - "HP", "HPOUT1L", - "HP", "HPOUT1R", - "SPK", "SPKOUT", - "SPKOUT", "HPOUT2L", - "SPKOUT", "HPOUT2R", - "Main Mic", "MICBIAS2", - "IN1R", "Main Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml new file mode 100644 index 000000000000..c9178d928ad4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml @@ -0,0 +1,99 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos SoC Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + i2s-controller: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: phandle of the I2S0. + - description: phandle of the I2S1. + + audio-codec: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: | + phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt; + - description: phandle of the HDMI transmitter node. + + audio-amplifier: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the MAX98504 amplifier. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + maxItems: 1 + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names for sources and sinks are: + the WM5110's pins: + "HPOUT1L", + "HPOUT1R", + "HPOUT2L", + "HPOUT2R", + "HPOUT3L", + "HPOUT3R", + MAX98504's pins: + "SPKOUT" + and the jacks on the board: + "HP", + "SPK", + "RCV", + "Main Mic", + "Sub Mic", + "Third Mic", + "Headset Mic". + +required: + - compatible + - model + - i2s-controller + - audio-codec + - audio-amplifier + - mic-bias-gpios + - samsung,audio-routing + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,tm2-audio"; + model = "wm5110"; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-codec = <&wm5110>, <&hdmi>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3>; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; + }; + -- 2.17.1

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Re: [alsa-devel] linux-next: Tree for Oct 17 (soundwire)
by Randy Dunlap 17 Oct '19

17 Oct '19

On 10/16/19 9:26 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20191016: > on x86_64: # CONFIG_BLOCK is not set CC [M] drivers/soundwire/intel_init.o In file included from ../drivers/soundwire/intel_init.c:12:0: ../include/linux/iomap.h: In function ‘iomap_sector’: ../include/linux/iomap.h:76:48: error: ‘SECTOR_SHIFT’ undeclared (first use in this function); did you mean ‘SECTIONS_SHIFT’? return (iomap->addr + pos - iomap->offset) >> SECTOR_SHIFT; ^~~~~~~~~~~~ SECTIONS_SHIFT -- ~Randy

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[alsa-devel] [PATCH] ALSA: hda - Force runtime PM on Nvidia HDMI codecs
by Lukas Wunner 17 Oct '19

17 Oct '19

Przemysław Kopa reports that since commit b516ea586d71 ("PCI: Enable NVIDIA HDA controllers"), the discrete GPU Nvidia GeForce GT 540M on his 2011 Samsung laptop refuses to runtime suspend, resulting in a power regression and excessive heat. Rivera Valdez witnesses the same issue with a GeForce GT 525M (GF108M) of the same era, as does another Arch Linux user named "R0AR" with a more recent GeForce GTX 1050 Ti (GP107M). The commit exposes the discrete GPU's HDA controller and all four codecs on the controller do not set the CLKSTOP and EPSS bits in the Supported Power States Response. They also do not set the PS-ClkStopOk bit in the Get Power State Response. hda_codec_runtime_suspend() therefore does not call snd_hdac_codec_link_down(), which prevents each codec and the PCI device from runtime suspending. The same issue is present on some AMD discrete GPUs and we addressed it by forcing runtime PM despite the bits not being set, see commit 57cb54e53bdd ("ALSA: hda - Force to link down at runtime suspend on ATI/AMD HDMI"). Do the same for Nvidia HDMI codecs. Fixes: b516ea586d71 ("PCI: Enable NVIDIA HDA controllers") Link: https://bbs.archlinux.org/viewtopic.php?pid=1865512 Link: https://bugs.freedesktop.org/show_bug.cgi?id=75985#c81 Reported-by: Przemysław Kopa <prymoo(a)gmail.com> Reported-by: Rivera Valdez <riveravaldez(a)ysinembargo.com> Signed-off-by: Lukas Wunner <lukas(a)wunner.de> Cc: Daniel Drake <dan(a)reactivated.net> Cc: stable(a)vger.kernel.org # v5.3+ --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index bca5de7..795cbda 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3474,6 +3474,8 @@ static int patch_nvhdmi(struct hda_codec *codec) nvhdmi_chmap_cea_alloc_validate_get_type; spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; + codec->link_down_at_suspend = 1; + generic_acomp_init(codec, &nvhdmi_audio_ops, nvhdmi_port2pin); return 0; -- 2.20.1

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[alsa-devel] [PATCH 1/7] ASoC: amd: No need PCI-MSI interrupts
by Ravulapati Vishnu vardhan rao 17 Oct '19

17 Oct '19

ACP-PCI controller driver does not depends msi interrupts. So removed msi related pci functions which have no use and does not impact on existing functionality. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec24..8f6bf00 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -46,14 +46,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, goto release_regions; } - /* check for msi interrupt support */ - ret = pci_enable_msi(pci); - if (ret) - /* msi is not enabled */ - irqflags = IRQF_SHARED; - else - /* msi is enabled */ - irqflags = 0; + irqflags = 0; addr = pci_resource_start(pci, 0); adata->acp3x_base = ioremap(addr, pci_resource_len(pci, 0)); @@ -112,7 +105,6 @@ static int snd_acp3x_probe(struct pci_dev *pci, return 0; unmap_mmio: - pci_disable_msi(pci); iounmap(adata->acp3x_base); release_regions: pci_release_regions(pci); @@ -129,7 +121,6 @@ static void snd_acp3x_remove(struct pci_dev *pci) platform_device_unregister(adata->pdev); iounmap(adata->acp3x_base); - pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); } -- 2.7.4

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[alsa-devel] [PATCH -next] ASoC: atmel: Fix build error
by YueHaibing 17 Oct '19

17 Oct '19

when do randbuilding, I got this error: sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': (.text+0x12f6): undefined reference to `atmel_pcm_pdc_platform_register' This is because SND_ATMEL_SOC_SSC_DMA=y, SND_ATMEL_SOC_SSC=y, but SND_ATMEL_SOC_SSC_PDC=m. Fix it bt reintroducing the default Kconfig. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/atmel/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c22..79e45f2 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -12,10 +12,14 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC tristate depends on HAS_DMA + default m if SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=m + default y if SND_ATMEL_SOC_SSC_PDC=y || (SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=y) config SND_ATMEL_SOC_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM + default m if SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=m + default y if SND_ATMEL_SOC_SSC_DMA=y || (SND_ATMEL_SOC_SSC_DMA=m && SND_ATMEL_SOC_SSC=y) config SND_ATMEL_SOC_SSC tristate -- 2.7.4

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[alsa-devel] [sound:for-next 2/20] drivers/pci/pci.c:5859:6: error: redefinition of 'pci_pr3_present'
by kbuild test robot 17 Oct '19

17 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: fa5a68b2abcefe2204075c80662e02cf59567a9f commit: 2e1a529c10e18e62da55eac42c278f11216dddaa [2/20] PCI: Add pci_pr3_present() helper to check Power Resource for D3hot config: parisc-c3000_defconfig (attached as .config) compiler: hppa-linux-gcc (GCC) 7.4.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 2e1a529c10e18e62da55eac42c278f11216dddaa # save the attached .config to linux build tree GCC_VERSION=7.4.0 make.cross ARCH=parisc If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): >> drivers/pci/pci.c:5859:6: error: redefinition of 'pci_pr3_present' bool pci_pr3_present(struct pci_dev *pdev) ^~~~~~~~~~~~~~~ In file included from include/linux/of_pci.h:5:0, from drivers/pci/pci.c:17: include/linux/pci.h:2317:13: note: previous definition of 'pci_pr3_present' was here static bool pci_pr3_present(struct pci_dev *pdev) { return false; } ^~~~~~~~~~~~~~~ drivers/pci/pci.c: In function 'pci_pr3_present': >> drivers/pci/pci.c:5870:13: error: dereferencing pointer to incomplete type 'struct acpi_device' return adev->power.flags.power_resources && ^~ >> drivers/pci/pci.c:5871:3: error: implicit declaration of function 'acpi_has_method'; did you mean 'acpi_has_watchdog'? [-Werror=implicit-function-declaration] acpi_has_method(adev->handle, "_PR3"); ^~~~~~~~~~~~~~~ acpi_has_watchdog In file included from include/linux/of_pci.h:5:0, from drivers/pci/pci.c:17: At top level: include/linux/pci.h:2317:13: warning: 'pci_pr3_present' defined but not used [-Wunused-function] static bool pci_pr3_present(struct pci_dev *pdev) { return false; } ^~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/pci_pr3_present +5859 drivers/pci/pci.c 5858 > 5859 bool pci_pr3_present(struct pci_dev *pdev) 5860 { 5861 struct acpi_device *adev; 5862 5863 if (acpi_disabled) 5864 return false; 5865 5866 adev = ACPI_COMPANION(&pdev->dev); 5867 if (!adev) 5868 return false; 5869 > 5870 return adev->power.flags.power_resources && > 5871 acpi_has_method(adev->handle, "_PR3"); 5872 } 5873 EXPORT_SYMBOL_GPL(pci_pr3_present); 5874 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ALSA: hda/realtek - Enable headset mic on Asus MJ401TA
by Daniel Drake 17 Oct '19

17 Oct '19

On Asus MJ401TA (with Realtek ALC256), the headset mic is connected to pin 0x19, with default configuration value 0x411111f0 (indicating no physical connection). Enable this by quirking the pin. Mic jack detection was also tested and found to be working. This enables use of the headset mic on this product. Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..e9fc5d8ff497 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5869,6 +5869,7 @@ enum { ALC225_FIXUP_WYSE_AUTO_MUTE, ALC225_FIXUP_WYSE_DISABLE_MIC_VREF, ALC286_FIXUP_ACER_AIO_HEADSET_MIC, + ALC256_FIXUP_ASUS_HEADSET_MIC, ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, @@ -6912,6 +6913,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC286_FIXUP_ACER_AIO_MIC_NO_PRESENCE }, + [ALC256_FIXUP_ASUS_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11020 }, /* headset mic with jack detect */ + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, [ALC256_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7108,6 +7118,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1a30, "ASUS X705UD", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), -- 2.20.1

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[alsa-devel] [PATCH v5 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 17 Oct '19

17 Oct '19

Hey, here's the 5th round for this series that adapts SOF to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is to unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. v5 changes: - Fixed a bug with codec power management at probe. To imitate snd_hda_intel driver, SOF needs to enable display power when doing codec probe for HDMI codecs. Affects patches 2 and 5. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) Kai Vehmanen (9): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: bxt-da7219-max98357a: common hdmi codec support ASoC: Intel: glk_rt5682_max98357a: common hdmi codec support ASoC: intel: sof_rt5682: common hdmi codec support ASoC: Intel: bxt_rt298: common hdmi codec support include/sound/hda_codec.h | 1 + include/sound/soc-acpi.h | 2 + sound/pci/hda/patch_hdmi.c | 19 ++- sound/soc/codecs/hdac_hda.c | 110 ++++++++++++++++-- sound/soc/codecs/hdac_hda.h | 13 ++- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 ++ sound/soc/intel/boards/bxt_rt298.c | 11 ++ sound/soc/intel/boards/glk_rt5682_max98357a.c | 11 ++ sound/soc/intel/boards/hda_dsp_common.h | 93 +++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.c | 10 +- sound/soc/intel/boards/skl_hda_dsp_common.h | 27 ++++- sound/soc/intel/boards/skl_hda_dsp_generic.c | 1 + sound/soc/intel/boards/sof_rt5682.c | 11 ++ sound/soc/sof/intel/Kconfig | 10 ++ sound/soc/sof/intel/hda-codec.c | 22 +++- sound/soc/sof/intel/hda.c | 6 + sound/soc/sof/intel/hda.h | 6 +- 17 files changed, 337 insertions(+), 27 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

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[alsa-devel] [PATCH 00/17] firewire: share the size of period for both playback and capture PCM substreams
by Takashi Sakamoto 17 Oct '19

17 Oct '19

Hi, This patchset is a preparation for enhancement of AMDTP domain that I addressed to my previous patchset: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/152430.html When queued packets for several IT/IR contexts in the same domain are handled in hardware IRQ context for an IT context, the number of events in the packets are mostly the same for each of contexts. This means that the size of period for PCM substreams associated to the contexts is also the same. At present, PCM substreams for the contexts can have own size of period. This commit adds a member into AMDTP domain to share the size of period for PCM substreams on AMDTP streams in the same domain. This patchset includes optimization to detect whether isochronous resources for AMDTP streams are reserved or not in pcm.open callback. Takashi Sakamoto (17): firewire-lib: add a member into AMDTP domain for events per period bebob: register the size of PCM period to AMDTP domain fireworks: register the size of PCM period to AMDTP domain oxfw: register the size of PCM period to AMDTP domain dice: register the size of PCM period to AMDTP domain firewire-digi00x: register the size of PCM period to AMDTP domain firewire-tascam: register the size of PCM period to AMDTP domain firewire-motu: register the size of PCM period to AMDTP domain fireface: register the size of PCM period to AMDTP domain bebob: use the same size of period for PCM substream in AMDTP streams fireworks: use the same size of period for PCM substream in AMDTP streams oxfw: use the same size of period for PCM substream in AMDTP streams dice: use the same size of period for PCM substream in AMDTP streams firewire-digi00x: use the same size of period for PCM substream in AMDTP streams firewire-tascam: use the same size of period for PCM substream in AMDTP streams firewire-motu: use the same size of period for PCM substream in AMDTP streams fireface: use the same size of period for PCM substreams in AMDTP streams sound/firewire/amdtp-stream.c | 4 ++ sound/firewire/amdtp-stream.h | 10 +++++ sound/firewire/bebob/bebob.h | 3 +- sound/firewire/bebob/bebob_midi.c | 2 +- sound/firewire/bebob/bebob_pcm.c | 43 ++++++++++++------ sound/firewire/bebob/bebob_stream.c | 11 ++++- sound/firewire/dice/dice-midi.c | 2 +- sound/firewire/dice/dice-pcm.c | 50 +++++++++++++++------ sound/firewire/dice/dice-stream.c | 8 +++- sound/firewire/dice/dice.h | 3 +- sound/firewire/digi00x/digi00x-midi.c | 2 +- sound/firewire/digi00x/digi00x-pcm.c | 39 ++++++++++++---- sound/firewire/digi00x/digi00x-stream.c | 11 ++++- sound/firewire/digi00x/digi00x.h | 3 +- sound/firewire/fireface/ff-pcm.c | 32 ++++++++++--- sound/firewire/fireface/ff-stream.c | 11 ++++- sound/firewire/fireface/ff.h | 3 +- sound/firewire/fireworks/fireworks.h | 3 +- sound/firewire/fireworks/fireworks_midi.c | 2 +- sound/firewire/fireworks/fireworks_pcm.c | 43 +++++++++++++----- sound/firewire/fireworks/fireworks_stream.c | 11 ++++- sound/firewire/motu/motu-midi.c | 2 +- sound/firewire/motu/motu-pcm.c | 33 ++++++++++---- sound/firewire/motu/motu-stream.c | 11 ++++- sound/firewire/motu/motu.h | 3 +- sound/firewire/oxfw/oxfw-midi.c | 4 +- sound/firewire/oxfw/oxfw-pcm.c | 45 +++++++++++++------ sound/firewire/oxfw/oxfw-stream.c | 12 ++++- sound/firewire/oxfw/oxfw.h | 3 +- sound/firewire/tascam/tascam-pcm.c | 38 ++++++++++++---- sound/firewire/tascam/tascam-stream.c | 11 ++++- sound/firewire/tascam/tascam.h | 3 +- 32 files changed, 354 insertions(+), 107 deletions(-) -- 2.20.1

2 18

[alsa-devel] [PATCH v4 1/2] PCI: Add a helper to check Power Resource Requirements _PR3 existence
by Kai-Heng Feng 17 Oct '19

17 Oct '19

A driver may want to know the existence of _PR3, to choose different runtime suspend behavior. A user will be add in next patch. This is mostly the same as nouveau_pr3_present(). Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v4: - Let caller to find its upstream port device. drivers/pci/pci.c | 16 ++++++++++++++++ include/linux/pci.h | 2 ++ 2 files changed, 18 insertions(+) diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index e7982af9a5d8..d03f624d8928 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -5856,6 +5856,22 @@ int pci_set_vga_state(struct pci_dev *dev, bool decode, return 0; } +bool pci_pr3_present(struct pci_dev *pdev) +{ + struct acpi_device *adev; + + if (acpi_disabled) + return false; + + adev = ACPI_COMPANION(&pdev->dev); + if (!adev) + return false; + + return adev->power.flags.power_resources && + acpi_has_method(adev->handle, "_PR3"); +} +EXPORT_SYMBOL_GPL(pci_pr3_present); + /** * pci_add_dma_alias - Add a DMA devfn alias for a device * @dev: the PCI device for which alias is added diff --git a/include/linux/pci.h b/include/linux/pci.h index f9088c89a534..1d15c5d49cdd 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2310,9 +2310,11 @@ struct irq_domain *pci_host_bridge_acpi_msi_domain(struct pci_bus *bus); void pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); +bool pci_pr3_present(struct pci_dev *pdev); #else static inline struct irq_domain * pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } +static bool pci_pr3_present(struct pci_dev *pdev) { return false; } #endif #ifdef CONFIG_EEH -- 2.17.1

3 5

[alsa-devel] [PATCH v5.3] ASoC: sgtl5000: add ADC mute control
by Richard Leitner 16 Oct '19

16 Oct '19

From: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> Upstream commit 631bc8f0134a ("ASoC: sgtl5000: Fix of unmute outputs on probe"), which is e9f621efaebd in v5.3 replaced snd_soc_component_write with snd_soc_component_update_bits and therefore no longer cleared the MUTE_ADC flag. This caused the ADC to stay muted and recording doesn't work any longer. This patch fixes this problem by adding a Switch control for MUTE_ADC. commit 694b14554d75 ("ASoC: sgtl5000: add ADC mute control") upstream This control mute/unmute the ADC input of SGTL5000 using its CHIP_ANA_CTRL register. Signed-off-by: Oleksandr Suvorov <oleksandr.suvorov(a)toradex.com> Reviewed-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Reviewed-by: Igor Opaniuk <igor.opaniuk(a)toradex.com> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Link: https://lore.kernel.org/r/20190719100524.23300-5-oleksandr.suvorov@toradex.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Richard Leitner <richard.leitner(a)skidata.com> Fixes: e9f621efaebd ("ASoC: sgtl5000: Fix of unmute outputs on probe") --- sound/soc/codecs/sgtl5000.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 3f28e7862b5b..b65232521ea8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -720,6 +720,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SGTL5000_CHIP_ANA_ADC_CTRL, 8, 1, 0, capture_6db_attenuate), SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), + SOC_SINGLE("Capture Switch", SGTL5000_CHIP_ANA_CTRL, 0, 1, 1), SOC_DOUBLE_TLV("Headphone Playback Volume", SGTL5000_CHIP_ANA_HP_CTRL, -- 2.21.0

2 1

[alsa-devel] [PATCH v6 0/7] ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 16 Oct '19

16 Oct '19

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, To be able to add support for the Allwinner H6 I've changed some of the original reg fields into function calls as this made it easier to setup for multi-channel audio especially across different SoCs. I've also stripped out all the other patches unrelated to this which I will deliver after support for the H6 has gone in. These other patches are required for HDMI audio which is driving this patchset and they can be found here https://github.com/codekipper/linux-sunxi/commits/upstream-i2s BR, CK --- v6 changes compared to v5 are: - rebased onto the recent tdm delivery - stripped out patches not required for the H6 delivery v5 changes compared to v4 are: - removed delivered patches. - Added more details to commit messages. - replaced some reg fields with function calls. - Added DSP_A and DSP_B support for H3 and later SoCs. - Added support for the Allwinner H6. v4 changes compared to v3 are: - Moved patches around so that the more controversial of patches are at the top of the stack. - Added more details to commit messages. - Fixed 20bit audio PCM format to use 4 bytes. - Reduced number of flags used to indicate a new SoC. v3 changes compared to v2 are: - added back slave mode changes - added back the use of tdm properties - changes to regmap and caching - removed loopback functionality - fixes to the channel offset mask v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. --- Jernej Skrabec (2): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S Marcus Cooper (5): ASoC: sun4i-i2s: Move channel select offset ASoC: sun4i-i2s: Add functions for RX and TX channel offsets ASoC: sun4i-i2s: Add functions for RX and TX channel enables ASoC: sun4i-i2s: Add functions for RX and TX channel selects ASoC: sun4i-i2s: Add functions for RX and TX channel mapping .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + sound/soc/sunxi/sun4i-i2s.c | 337 ++++++++++++++++-- 2 files changed, 305 insertions(+), 34 deletions(-) -- 2.23.0

3 13

[alsa-devel] [PATCH] ASoC: wm8958: use <asm/unaligned.h> to simplify code
by Ben Dooks (Codethink) 16 Oct '19

16 Oct '19

Simplify the memcpy/be32_to_cpu() code by simply using get_unaligned_be32() throughout and makes the code nicer to look at. This fixes the following warnings from sparse: sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:62:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:69:15: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:72:18: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:91:17: warning: cast to restricted __be64 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:108:29: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 sound/soc/codecs/wm8958-dsp2.c:120:26: warning: cast to restricted __be32 Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: patches(a)opensource.cirrus.com Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/wm8958-dsp2.c | 22 +++++++++------------- 1 file changed, 9 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 18535b326680..ca42445b649d 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -25,6 +25,8 @@ #include <linux/mfd/wm8994/pdata.h> #include <linux/mfd/wm8994/gpio.h> +#include <asm/unaligned.h> + #include "wm8994.h" #define WM_FW_BLOCK_INFO 0xff @@ -58,18 +60,15 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, } if (memcmp(fw->data, "WMFW", 4) != 0) { - memcpy(&data32, fw->data, sizeof(data32)); - data32 = be32_to_cpu(data32); + data32 = get_unaligned_be32(fw->data); dev_err(component->dev, "%s: firmware has bad file magic %08x\n", name, data32); goto err; } - memcpy(&data32, fw->data + 4, sizeof(data32)); - len = be32_to_cpu(data32); + len = get_unaligned_be32(fw->data + 4); + data32 = get_unaligned_be32(fw->data + 8); - memcpy(&data32, fw->data + 8, sizeof(data32)); - data32 = be32_to_cpu(data32); if ((data32 >> 24) & 0xff) { dev_err(component->dev, "%s: unsupported firmware version %d\n", name, (data32 >> 24) & 0xff); @@ -87,9 +86,8 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, } if (check) { - memcpy(&data64, fw->data + 24, sizeof(u64)); - dev_info(component->dev, "%s timestamp %llx\n", - name, be64_to_cpu(data64)); + data64 = get_unaligned_be64(fw->data + 24); + dev_info(component->dev, "%s timestamp %llx\n", name, data64); } else { snd_soc_component_write(component, 0x102, 0x2); snd_soc_component_write(component, 0x900, 0x2); @@ -104,8 +102,7 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, goto err; } - memcpy(&data32, data + 4, sizeof(data32)); - block_len = be32_to_cpu(data32); + block_len = get_unaligned_be32(data + 4); if (block_len + 8 > len) { dev_err(component->dev, "%zd byte block longer than file\n", block_len); @@ -116,8 +113,7 @@ static int wm8958_dsp2_fw(struct snd_soc_component *component, const char *name, goto err; } - memcpy(&data32, data, sizeof(data32)); - data32 = be32_to_cpu(data32); + data32 = get_unaligned_be32(data); switch ((data32 >> 24) & 0xff) { case WM_FW_BLOCK_INFO: -- 2.23.0

3 2

[alsa-devel] [PATCH] ASoC: dt-bindings: rt1011: add r0 and temperature device property
by shumingf＠realtek.com 16 Oct '19

16 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> Typically, the r0 (calibration data) and temperature were measured in the factory. This information is written into the non-volatile area where keeps data whether factory reset or OS update. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt1011.txt | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt1011.txt b/Documentation/devicetree/bindings/sound/rt1011.txt index 35a23e60d679..02d53b9aa247 100644 --- a/Documentation/devicetree/bindings/sound/rt1011.txt +++ b/Documentation/devicetree/bindings/sound/rt1011.txt @@ -20,6 +20,14 @@ Required properties: | 1 | 1 | 0x3b | ------------------------------------- +Optional properties: + +- realtek,temperature_calib + u32. The temperature was measured while doing the calibration. Units: Celsius degree + +- realtek,r0_calib + u32. This is r0 calibration data which was measured in factory mode. + Pins on the device (for linking into audio routes) for RT1011: * SPO @@ -29,4 +37,6 @@ Example: rt1011: codec@38 { compatible = "realtek,rt1011"; reg = <0x38>; + realtek,temperature_calib = <25>; + realtek,r0_calib = <0x224050>; }; -- 2.23.0

2 3

[alsa-devel] [PATCH v2] ASoC: rt1011: set tx/rx slots from tx/rx_mask in TDM case
by shumingf＠realtek.com 16 Oct '19

16 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> The TX/RX slot configuration use tx/rx_mask which requested by the machine driver. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 142 ++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt1011.h | 23 ++++++ 2 files changed, 160 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index be1e276e3631..b927e61b32df 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1805,8 +1805,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, struct snd_soc_component *component = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - unsigned int val = 0, tdm_en = 0; - int ret = 0; + unsigned int val = 0, tdm_en = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit, last_bit; snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) @@ -1829,6 +1829,7 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, break; default: ret = -EINVAL; + goto _set_tdm_err_; } switch (slot_width) { @@ -1848,6 +1849,135 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, break; default: ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + first_bit = find_next_bit((unsigned long *)&rx_mask, 32, 0); + if (rx_slotnum > 1 || rx_slotnum == 0) { + ret = -EINVAL; + dev_dbg(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_L_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + case 1: + case 3: + case 5: + case 7: + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_R_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + break; + default: + ret = -EINVAL; + goto _set_tdm_err_; + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + first_bit = find_next_bit((unsigned long *)&tx_mask, 32, 0); + last_bit = find_last_bit((unsigned long *)&tx_mask, 32); + if (tx_slotnum > 2 || (last_bit-first_bit) > 1) { + ret = -EINVAL; + dev_dbg(component->dev, "too many tx slots or tx slot location error\n"); + goto _set_tdm_err_; + } + + if (tx_slotnum == 1) { + snd_soc_component_update_bits(component, RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, first_bit); + switch (first_bit) { + case 1: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC1_1_MASK, + RT1011_TDM_I2S_RX_ADC1_1_LL); + break; + case 3: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC2_1_MASK, + RT1011_TDM_I2S_RX_ADC2_1_LL); + break; + case 5: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC3_1_MASK, + RT1011_TDM_I2S_RX_ADC3_1_LL); + break; + case 7: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC4_1_MASK, + RT1011_TDM_I2S_RX_ADC4_1_LL); + break; + case 0: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC1_1_MASK, 0); + break; + case 2: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC2_1_MASK, 0); + break; + case 4: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC3_1_MASK, 0); + break; + case 6: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC4_1_MASK, 0); + break; + default: + ret = -EINVAL; + dev_dbg(component->dev, + "tx slot location error\n"); + goto _set_tdm_err_; + } + } else if (tx_slotnum == 2) { + switch (first_bit) { + case 0: + case 2: + case 4: + case 6: + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, + RT1011_TDM_I2S_DOCK_ADCDAT_2CH | first_bit); + break; + default: + ret = -EINVAL; + dev_dbg(component->dev, + "tx slot location should be paired and start from slot0/2/4/6\n"); + goto _set_tdm_err_; + } } snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, @@ -1860,10 +1990,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); - snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, - RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, - RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); + if (tx_slotnum) + snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, + RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, + RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); +_set_tdm_err_: snd_soc_dapm_mutex_unlock(dapm); return ret; } diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 2d65983f3d0f..3f7dea1191ad 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -460,6 +460,23 @@ #define RT1011_TDM_I2S_DOCK_EN_1_MASK (0x1 << 3) #define RT1011_TDM_I2S_DOCK_EN_1_SFT 3 #define RT1011_TDM_I2S_DOCK_EN_1 (0x1 << 3) +#define RT1011_TDM_ADCDAT1_DATA_LOCATION (0x7 << 0) + +/* TDM1 Setting-3 (0x0118) */ +#define RT1011_TDM_I2S_RX_ADC1_1_MASK (0x3 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_MASK (0x3 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_MASK (0x3 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_MASK (0x3 << 0) +#define RT1011_TDM_I2S_RX_ADC1_1_LL (0x2 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_LL (0x2 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_LL (0x2 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_LL (0x2 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1011_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1011_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1011_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1011_TDM_I2S_TX_R_DAC1_1_SFT 8 /* TDM2 Setting-2 (0x0120) */ #define RT1011_TDM_I2S_DOCK_ADCDAT_LEN_2_MASK (0x7 << 13) @@ -585,6 +602,12 @@ #define RT1011_STP_T0_EN_BIT 6 #define RT1011_STP_T0_EN (0x1 << 6) +/* Cross Biquad Setting-1 (0x0702) */ +#define RT1011_MONO_LR_SEL_MASK (0x3 << 5) +#define RT1011_MONO_L_CHANNEL (0x0 << 5) +#define RT1011_MONO_R_CHANNEL (0x1 << 5) +#define RT1011_MONO_LR_MIX_CHANNEL (0x2 << 5) + /* ClassD Internal Setting-1 (0x1300) */ #define RT1011_DRIVER_READY_SPK (0x1 << 12) #define RT1011_DRIVER_READY_SPK_BIT 12 -- 2.23.0

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[alsa-devel] [PATCH] ASoC: rt1011: Read and apply r0 and temperature device property
by shumingf＠realtek.com 16 Oct '19

16 Oct '19

From: Shuming Fan <shumingf(a)realtek.com> Typically, the r0 (calibration data) and temperature were measured in the factory. This information is written into the non-volatile area where keeps data whether factory reset or OS update. In Chromium OS case, the coreboot will read the info from VPD and create the device property for each rt1011. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 41 ++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/rt1011.h | 1 + 2 files changed, 41 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index b927e61b32df..ad049cfddcb0 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -2328,8 +2328,12 @@ static void rt1011_calibration_work(struct work_struct *work) struct rt1011_priv *rt1011 = container_of(work, struct rt1011_priv, cali_work); struct snd_soc_component *component = rt1011->component; + unsigned int r0_integer, r0_factor, format; - rt1011_calibrate(rt1011, 1); + if (rt1011->r0_calib) + rt1011_calibrate(rt1011, 0); + else + rt1011_calibrate(rt1011, 1); /* * This flag should reset after booting. @@ -2340,6 +2344,39 @@ static void rt1011_calibration_work(struct work_struct *work) /* initial */ rt1011_reg_init(component); + + /* Apply temperature and calibration data from device property */ + if (rt1011->temperature_calib) { + snd_soc_component_update_bits(component, + RT1011_STP_INITIAL_RESISTANCE_TEMP, 0x3ff, + (rt1011->temperature_calib << 2)); + } + + if (rt1011->r0_calib) { + rt1011->r0_reg = rt1011->r0_calib; + + format = 2147483648U; /* 2^24 * 128 */ + r0_integer = format / rt1011->r0_reg / 128; + r0_factor = ((format / rt1011->r0_reg * 100) / 128) + - (r0_integer * 100); + dev_info(component->dev, "DP r0 resistance about %d.%02d ohm, reg=0x%X\n", + r0_integer, r0_factor, rt1011->r0_reg); + + rt1011_r0_load(rt1011); + } +} + +static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device *dev) +{ + device_property_read_u32(dev, "realtek,temperature_calib", + &rt1011->temperature_calib); + device_property_read_u32(dev, "realtek,r0_calib", + &rt1011->r0_calib); + + dev_dbg(dev, "%s: r0_calib: 0x%x, temperture_calib: 0x%x", + __func__, rt1011->r0_calib, rt1011->temperature_calib); + + return 0; } static int rt1011_i2c_probe(struct i2c_client *i2c, @@ -2356,6 +2393,8 @@ static int rt1011_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt1011); + rt1011_parse_dp(rt1011, &i2c->dev); + rt1011->regmap = devm_regmap_init_i2c(i2c, &rt1011_regmap); if (IS_ERR(rt1011->regmap)) { ret = PTR_ERR(rt1011->regmap); diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 3f7dea1191ad..68fadc15fa8c 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -690,6 +690,7 @@ struct rt1011_priv { int bq_drc_set; unsigned int r0_reg, cali_done; + unsigned int r0_calib, temperature_calib; int recv_spk_mode; }; -- 2.23.0

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[alsa-devel] [PATCH] ASoC: sof: include types.h at header.h
by Kuninori Morimoto 16 Oct '19

16 Oct '19

Content-Transfer-Encoding: 8bit Without <types.h> we will get these error linux/include/sound/sof/header.h:125:2: error: unknown type name ‘uint32_t’uint32_t size; linux/include/sound/sof/header.h:136:2: error: unknown type name ‘uint32_t’uint32_t size; linux/include/sound/sof/header.h:137:2: error: unknown type name ‘uint32_t’uint32_t cmd; ... linux/include/sound/sof/dai-imx.h:18:2: error: unknown type name ‘uint16_t’uint16_t reserved1; linux/include/sound/sof/dai-imx.h:30:2: error: unknown type name ‘uint16_t’uint16_t tdm_slot_width; linux/include/sound/sof/dai-imx.h:31:2: error: unknown type name ‘uint16_t’uint16_t reserved2; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/sof/header.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index 10f00c0..332143f 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -9,6 +9,7 @@ #ifndef __INCLUDE_SOUND_SOF_HEADER_H__ #define __INCLUDE_SOUND_SOF_HEADER_H__ +#include <linux/types.h> #include <uapi/sound/sof/abi.h> /** \addtogroup sof_uapi uAPI -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: rt1011: Read and apply r0 and temperature device property
by Mark Brown 16 Oct '19

16 Oct '19

On Wed, Oct 16, 2019 at 11:28:47AM +0000, Shuming [范書銘] wrote: > > These are new DT bindings and need to be added to the binding document. > Sure, could I send another patch for it? Yes, the DT bindings should go in a separate patch anyway.

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Re: [alsa-devel] [PATCH] ASoC: rt1011: Read and apply r0 and temperature device property
by Cheng-yi Chiang 16 Oct '19

16 Oct '19

On Wed, Oct 16, 2019 at 7:29 PM Shuming [范書銘] <shumingf(a)realtek.com> wrote: > > > On Wed, Oct 16, 2019 at 04:58:45PM +0800, shumingf(a)realtek.com wrote: > > > > > +static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device > > > +*dev) { > > > + device_property_read_u32(dev, "realtek,temperature_calib", > > > + &rt1011->temperature_calib); > > > + device_property_read_u32(dev, "realtek,r0_calib", > > > + &rt1011->r0_calib); > > > > These are new DT bindings and need to be added to the binding document. > > Sure, could I send another patch for it? > > > ------Please consider the environment before printing this e-mail. Hi Shuming, Yes, you will need a patch before this one to add the binding documentation. Thanks!

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[alsa-devel] [PATCH 0/2] soundwire: Kconfig/build improvements
by Pierre-Louis Bossart 16 Oct '19

16 Oct '19

The Intel kbuild test reported issues with COMPILE_TEST or cross-compilation when SOF is enabled, fix. This has no functional impact. Pierre-Louis Bossart (2): soundwire: intel: add missing headers for cross-compilation soundwire: intel: remove X86 dependency drivers/soundwire/Kconfig | 2 +- drivers/soundwire/intel.c | 1 + drivers/soundwire/intel_init.c | 1 + 3 files changed, 3 insertions(+), 1 deletion(-) -- 2.20.1

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[alsa-devel] [PATCH 0/2] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 16 Oct '19

16 Oct '19

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the for-next. v3 changes: 1. https://patchwork.kernel.org/patch/11164283/ and https://patchwork.kernel.org/patch/11164305/ has been merged. 2. change the name of mtk_wdt_compatible to mtk_wdt_data 3. remove toprgu_reset struct and use mtk_wdt_dev instead 4. get the value of sw_rst_num from .h file 5. adddd mt2712-resets.h for mt2712 6. improve commit message v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on 3. patchwork list: https://patchwork.kernel.org/cover/11164285/ https://patchwork.kernel.org/patch/11164295/ https://patchwork.kernel.org/patch/11164299/ https://patchwork.kernel.org/patch/11164283/ https://patchwork.kernel.org/patch/11164305/ v1 changes: 1. patchwork list: https://patchwork.kernel.org/cover/11164173/ https://patchwork.kernel.org/patch/11164181/ https://patchwork.kernel.org/patch/11164185/ https://patchwork.kernel.org/patch/11164187/ https://patchwork.kernel.org/patch/11164175/ yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../devicetree/bindings/watchdog/mtk-wdt.txt | 10 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 111 +++++++++++++++++- .../reset-controller/mt2712-resets.h | 22 ++++ .../reset-controller/mt8183-resets.h | 15 +++ 5 files changed, 155 insertions(+), 4 deletions(-) create mode 100644 include/dt-bindings/reset-controller/mt2712-resets.h -- 2.18.0

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[alsa-devel] [PATCH] ASoC: pcm3168a: Use fixup instead of constraint for channels and formats
by Peter Ujfalusi 16 Oct '19

16 Oct '19

The snd_pcm_hw_constraint_minmax() works fine when a single codec is connected to a single CPU DAI, but in multicodec or DPCM setup the constraints placed by the driver will apply to the whole PCM stream (FE included) and thus prevents more than 8 playback channels for example. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have a board which uses two pcm3168a codec connected to one McASP port with parallel serializers. One 16 channel (2x 8 channel) playback and one 12 (2x 6 channel) capture should be possible as I use only one CPU DAI + platform. However no matter how I tried to set things up ALSA blocked more than 8 channel playback in simple multicodec or DPCM setup. It turned out that the constraint set by the codec on the BE is actually applies to the FE as well as they both use the same runtime. The codec still have different constraints depending on the interface format, so I replaced the constraint with a fixup function to make sure that the codec will not advertise non supported formats. Is this a know feature? Another thing which I have came across is the lack of support for parallel data pins in ASoC core. For example the soc-pcm only checks the tdm slots to fix up the channels supported by the codec, but when we have multiple pins in use and they are operate in parallel, then the number of slots must be multiplied by the data pins in use. I have wip patches for this, adding int snd_soc_dai_set_data_pins(struct snd_soc_dai *dai, unsigned int tx_data_pins, unsigned int rx_data_pins); and then using the information in soc-pcm, but need to check further to see if there are other places this might be needed. Regards, Peter sound/soc/codecs/pcm3168a.c | 98 +++++++++++++++---------------------- 1 file changed, 40 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 88b75695fbf7..708dac27feff 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -62,6 +62,7 @@ struct pcm3168a_priv { unsigned long sysclk; struct pcm3168a_io_params io_params[2]; + struct snd_soc_dai_driver dai_drv[2]; }; static const char *const pcm3168a_roll_off[] = { "Sharp", "Slow" }; @@ -314,6 +315,37 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } +static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); + u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; + unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; + + if (!pcm3168a->io_params[dai->id].fmt) + return; + + if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { + /* S16_LE is only supported in RIGHT_J mode */ + formats |= SNDRV_PCM_FMTBIT_S16_LE; + + /* + * If multi DIN/DOUT is not selected, RIGHT_J can only support + * two channels (no TDM support) + */ + if (pcm3168a->io_params[dai->id].tdm_slots != 2) + channel_max = 2; + } + + if (dai->id == PCM3168A_DAI_DAC) { + dai->driver->playback.channels_max = channel_max; + dai->driver->playback.formats = formats; + } else { + dai->driver->capture.channels_max = channel_max; + dai->driver->capture.formats = formats; + } +} + static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format) { struct snd_soc_component *component = dai->component; @@ -376,6 +408,8 @@ static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format) regmap_update_bits(pcm3168a->regmap, reg, mask, fmt << shift); + pcm3168a_update_fixup_pcm_stream(dai); + return 0; } @@ -409,6 +443,8 @@ static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, else io_params->tdm_mask = rx_mask; + pcm3168a_update_fixup_pcm_stream(dai); + return 0; } @@ -530,63 +566,7 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pcm3168a_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); - unsigned int sample_min; - unsigned int channel_max; - unsigned int channel_maxs[] = { - 8, /* DAC */ - 6 /* ADC */ - }; - - /* - * Available Data Bits - * - * RIGHT_J : 24 / 16 - * LEFT_J : 24 - * I2S : 24 - * - * TDM available - * - * I2S - * LEFT_J - */ - switch (pcm3168a->io_params[dai->id].fmt) { - case PCM3168A_FMT_RIGHT_J: - sample_min = 16; - channel_max = 2; - break; - case PCM3168A_FMT_LEFT_J: - case PCM3168A_FMT_I2S: - case PCM3168A_FMT_DSP_A: - case PCM3168A_FMT_DSP_B: - sample_min = 24; - channel_max = channel_maxs[dai->id]; - break; - default: - sample_min = 24; - channel_max = 2; - } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - sample_min, 32); - - /* Allow all channels in multi DIN/DOUT mode */ - if (pcm3168a->io_params[dai->id].tdm_slots == 2) - channel_max = channel_maxs[dai->id]; - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - 2, channel_max); - - return 0; -} static const struct snd_soc_dai_ops pcm3168a_dai_ops = { - .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params, @@ -776,8 +756,10 @@ int pcm3168a_probe(struct device *dev, struct regmap *regmap) pm_runtime_enable(dev); pm_runtime_idle(dev); - ret = devm_snd_soc_register_component(dev, &pcm3168a_driver, pcm3168a_dais, - ARRAY_SIZE(pcm3168a_dais)); + memcpy(pcm3168a->dai_drv, pcm3168a_dais, sizeof(pcm3168a->dai_drv)); + ret = devm_snd_soc_register_component(dev, &pcm3168a_driver, + pcm3168a->dai_drv, + ARRAY_SIZE(pcm3168a->dai_drv)); if (ret) { dev_err(dev, "failed to register component: %d\n", ret); goto err_regulator; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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Re: [alsa-devel] Focusrite Scarlett 18i8 3rd Gen audio interface (Re: Alsa-devel Digest, Vol 152, Issue 150)
by Hin-Tak Leung 15 Oct '19

15 Oct '19

> Date: Tue, 15 Oct 2019 00:10:28 +1030 > From: "Geoffrey D. Bennett" <g(a)b4.vu> > To: wwp <subscript(a)free.fr> > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Focusrite Scarlett 18i8 3rd Gen audio > interface > Message-ID: <20191014134028.GA11018(a)b4.vu> > Content-Type: text/plain; charset=us-ascii > Hi wwp, > On Sun, Oct 13, 2019 at 10:20:30AM +0200, wwp wrote: > > Hello, > > > > I've recently bought a Focusrite Scarlett 18i8 3rd Gen audio interface, > > great device, but in CentOS 7, with either the available 3.x or 4.x > > kernels, > ... > > Of course, adding the 0x1235:8214 USB ID to the quirks just doesn't > > work. I presume there's more work like the patch submitted here (I saw > > the archives) in the "[PATCH RFC V2] ALSA: usb-audio: Scarlett Gen 2 > > mixer interface" mail thread? Have you tried treating it as a 18i8 2nd Gen device first, at least? FWIW, I had adapted Geoffrey's work as a DKMS module: https://github.com/HinTak/sound-usb-dkms But you'll still need to be running as recent as a 5.2.x release kernel to use it. It just makes it easier to play with the code without recompiling the whole kernel. HTH. 3.x and 4.x is definitely way too old!

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[alsa-devel] Focusrite Scarlett 18i8 3rd Gen audio interface
by wwp 15 Oct '19

15 Oct '19

Hello, I've recently bought a Focusrite Scarlett 18i8 3rd Gen audio interface, great device, but in CentOS 7, with either the available 3.x or 4.x kernels, the device just appears in the available Alsa devices, but no mixer is available, the device direct hardware monitor output is also not working (it is, by default, and works well in Windows, but I presume that w/ no mixer support in Alsa, it's expected that nothing works). I'm able to bound I/O ports in QJackCtl but nothing goes out of the device. Of course, adding the 0x1235:8214 USB ID to the quirks just doesn't work. I presume there's more work like the patch submitted here (I saw the archives) in the "[PATCH RFC V2] ALSA: usb-audio: Scarlett Gen 2 mixer interface" mail thread? Do you it is worth trying applying that 2nd Gen patch to one of the v4 kernel available for CentOS 7, adding the 3rd Gen quirks and see how it behaves? Or is there already progress on supporting the 3rd Gen one too? Regards, -- wwp https://useplaintext.email/

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