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

[alsa-devel] [PATCH] ASoC: fsl_asrc: refine the setting of internal clock divider
by Shengjiu Wang 18 Oct '19

18 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

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