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[alsa-devel] [PATCH v7 0/2] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 10 Oct '19

10 Oct '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 and Lenovo YOGA C630 Laptop based on SDM850 with WSA8815 speaker amplifiers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v6: - Added reviewed-by tag from Pierre and Rob - Removed udelay argument as suggested by Pierre. - Added device id for WSA8815 - rebased on sound-next Srinivas Kandagatla (2): dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 62 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1113 +++++++++++++++++ 4 files changed, 1187 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

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[alsa-devel] [PATCH] ASoC: msm8916-wcd-digital: add missing MIX2 path for RX1/2
by Srinivas Kandagatla 09 Oct '19

09 Oct '19

This patch adds missing MIX2 path on RX1/2 which take IIR1 and IIR2 as inputs. Without this patch sound card fails to intialize with below warning: ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX2 MIX2 INP1 ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX2 MIX2 INP1 Reported-by: Stephan Gerhold <stephan(a)gerhold.net> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Tested-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/codecs/msm8916-wcd-digital.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 9fa5d44fdc79..58b2468fb2a7 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -243,6 +243,10 @@ static const char *const rx_mix1_text[] = { "ZERO", "IIR1", "IIR2", "RX1", "RX2", "RX3" }; +static const char * const rx_mix2_text[] = { + "ZERO", "IIR1", "IIR2" +}; + static const char *const dec_mux_text[] = { "ZERO", "ADC1", "ADC2", "ADC3", "DMIC1", "DMIC2" }; @@ -270,6 +274,16 @@ static const struct soc_enum rx3_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B2_CTL, 0, 6, rx_mix1_text), }; +/* RX1 MIX2 */ +static const struct soc_enum rx_mix2_inp1_chain_enum = + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX1_B3_CTL, + 0, 3, rx_mix2_text); + +/* RX2 MIX2 */ +static const struct soc_enum rx2_mix2_inp1_chain_enum = + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B3_CTL, + 0, 3, rx_mix2_text); + /* DEC */ static const struct soc_enum dec1_mux_enum = SOC_ENUM_SINGLE( LPASS_CDC_CONN_TX_B1_CTL, 0, 6, dec_mux_text); @@ -309,6 +323,10 @@ static const struct snd_kcontrol_new rx3_mix1_inp2_mux = SOC_DAPM_ENUM( "RX3 MIX1 INP2 Mux", rx3_mix1_inp_enum[1]); static const struct snd_kcontrol_new rx3_mix1_inp3_mux = SOC_DAPM_ENUM( "RX3 MIX1 INP3 Mux", rx3_mix1_inp_enum[2]); +static const struct snd_kcontrol_new rx1_mix2_inp1_mux = SOC_DAPM_ENUM( + "RX1 MIX2 INP1 Mux", rx_mix2_inp1_chain_enum); +static const struct snd_kcontrol_new rx2_mix2_inp1_mux = SOC_DAPM_ENUM( + "RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum); /* Digital Gain control -38.4 dB to +38.4 dB in 0.3 dB steps */ static const DECLARE_TLV_DB_SCALE(digital_gain, -3840, 30, 0); @@ -740,6 +758,10 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = { &rx3_mix1_inp2_mux), SND_SOC_DAPM_MUX("RX3 MIX1 INP3", SND_SOC_NOPM, 0, 0, &rx3_mix1_inp3_mux), + SND_SOC_DAPM_MUX("RX1 MIX2 INP1", SND_SOC_NOPM, 0, 0, + &rx1_mix2_inp1_mux), + SND_SOC_DAPM_MUX("RX2 MIX2 INP1", SND_SOC_NOPM, 0, 0, + &rx2_mix2_inp1_mux), SND_SOC_DAPM_MUX("CIC1 MUX", SND_SOC_NOPM, 0, 0, &cic1_mux), SND_SOC_DAPM_MUX("CIC2 MUX", SND_SOC_NOPM, 0, 0, &cic2_mux), -- 2.21.0

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[alsa-devel] [PATCH 0/2] ASoC: simple_card_utils.h: Fix two potential compilation errors
by Daniel Baluta 09 Oct '19

09 Oct '19

For this to happen we symbol DEBUG to be defined. Daniel Baluta (2): ASoC: simple_card_utils.h: Add missing include ASoC: Fix potential multiple redefinition error include/sound/simple_card_utils.h | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) -- 2.17.1

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[alsa-devel] [RFC PATCH] ASoC: soc-dapm: Invalidate DAPM path during dapm addition of routes
by Srinivas Kandagatla 09 Oct '19

09 Oct '19

From: Sudheer Papothi <spapothi(a)codeaurora.org> During sound card registration, dapm adds routes of codec and other component paths, but the invalidation of the widgets in these paths will happen only when the sound card is instantiated. As these routes are added before sound card instantiation, these widgets are not invalidated until a playback or recording usecase is started. Audio playback or recording usecase is not started in the case of codec loopback. So, if codec loopback is performed just after soundcard registration, then the widgets are not powered up as those widgets are not invalidated, results into codec loopback failure. Change is to remove the sound card instantiation check condition in dapm add paths, so widgets get invalidated whenever they are added. Signed-off-by: Sudheer Papothi <spapothi(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-dapm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b6378f025836..13544f7c850b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2862,8 +2862,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(widgets[dir], "Route added"); } - if (dapm->card->instantiated && path->connect) - dapm_path_invalidate(path); + dapm_path_invalidate(path); return 0; err: -- 2.21.0

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[alsa-devel] [RFC PATCH] ASoC: soc-dapm: Skip suspending widgets with ignore flag
by Srinivas Kandagatla 09 Oct '19

09 Oct '19

From: Sudheer Papothi <spapothi(a)codeaurora.org> For wigdets which have set the suspend ignore flag asoc framework shouldn't mark them as dirty when ASoC suspend function is called. This change adds check to skip suspending the widgets with the flag set. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Sudheer Papothi <spapothi(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 13544f7c850b..4ecfd32e59b8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -303,6 +303,8 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card) mutex_lock(&card->dapm_mutex); list_for_each_entry(w, &card->widgets, list) { + if (w->ignore_suspend) + continue; if (w->is_ep) { dapm_mark_dirty(w, "Rechecking endpoints"); if (w->is_ep & SND_SOC_DAPM_EP_SINK) -- 2.21.0

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[alsa-devel] [PATCH 0/1] [for 4.14] VAG power control improvement for sgtl5000 codec
by Oleksandr Suvorov 09 Oct '19

09 Oct '19

This backport is for 4.14-stable. Oleksandr Suvorov (1): ASoC: sgtl5000: Improve VAG power and mute control sound/soc/codecs/sgtl5000.c | 234 +++++++++++++++++++++++++++++++----- 1 file changed, 203 insertions(+), 31 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH 0/1] [for 4.4/4.9] VAG power control improvement for sgtl5000 codec
by Oleksandr Suvorov 09 Oct '19

09 Oct '19

This is a backport to stable kernel versions 4.4 and 4.9. Oleksandr Suvorov (1): ASoC: sgtl5000: Improve VAG power and mute control sound/soc/codecs/sgtl5000.c | 234 +++++++++++++++++++++++++++++++----- 1 file changed, 203 insertions(+), 31 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: msm8916-wcd-digital: Remove broken MIX2 DAPM routes
by Stephan Gerhold 09 Oct '19

09 Oct '19

Since 5.3-rc1, the kernel (correctly) complains about missing widgets when loading msm8916-wcd-digital: ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX2 MIX2 INP1 ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX2 MIX2 INP1 This happens because RX1/2 MIX2 INP1 is not actually supported by msm8916-wcd-digital for some reason. There were some traces of it in the initial patch, but the last remnants were removed in commit fecd5c09ddf8 ("ASoC: msm8916-wcd-digital: remove few unused variables"). On 5.3 everything seemingly worked even with that error, but on 5.4-rc1 the sound card is no longer successfully registered: qcom-apq8016-sbc 7702000.sound: ASoC: failed to instantiate card -19 This happens because a check for the return code of snd_soc_dapm_add_routes() was added in commit daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()"). Avoid the error by removing the routes that reference the missing widgets. Fixes: ef8a4757a6db ("ASoC: msm8916-wcd-digital: Add sidetone support") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/codecs/msm8916-wcd-digital.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 9fa5d44fdc79..36d8f619e298 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1022,11 +1022,6 @@ static const struct snd_soc_dapm_route msm8916_wcd_digital_audio_map[] = { {"RX3 MIX1 INP2", "IIR1", "IIR1"}, {"RX3 MIX1 INP2", "IIR2", "IIR2"}, - {"RX1 MIX2 INP1", "IIR1", "IIR1"}, - {"RX2 MIX2 INP1", "IIR1", "IIR1"}, - {"RX1 MIX2 INP1", "IIR2", "IIR2"}, - {"RX2 MIX2 INP1", "IIR2", "IIR2"}, - {"IIR1", NULL, "IIR1 INP1 MUX"}, {"IIR1 INP1 MUX", "DEC1", "DEC1 MUX"}, {"IIR1 INP1 MUX", "DEC2", "DEC2 MUX"}, -- 2.23.0

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[alsa-devel] [PATCH] ASoC: jz4740: Remove unused match variable
by Stephen Boyd 09 Oct '19

09 Oct '19

After commit 67ad656bdd70 ("ASoC: jz4740: Use of_device_get_match_data()"), the match local variable is unused and the compiler rightly warns. sound/soc/jz4740/jz4740-i2s.c: In function 'jz4740_i2s_dev_probe': sound/soc/jz4740/jz4740-i2s.c:500:29: warning: unused variable 'match' [-Wunused-variable] 500 | const struct of_device_id *match; Drop it. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Paul Cercueil <paul(a)crapouillou.net> 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: Rob Herring <robh+dt(a)kernel.org> Cc: Frank Rowand <frowand.list(a)gmail.com> Cc: <alsa-devel(a)alsa-project.org> Fixes: 67ad656bdd70 ("ASoC: jz4740: Use of_device_get_match_data()") Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- Can also be squashed. Sorry I missed this warning in the compile log. sound/soc/jz4740/jz4740-i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index d2dab4d24b87..38d48d101783 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -497,7 +497,6 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) struct jz4740_i2s *i2s; struct resource *mem; int ret; - const struct of_device_id *match; i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) base-commit: 442630f691a1537b7e0cc35e3d580222077549cb -- Sent by a computer through tubes

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[alsa-devel] [PATCH v3] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 09 Oct '19

09 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems. The examples are taken from the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 34 +++++ sound/hda/Kconfig | 10 +- sound/hda/Makefile | 7 +- sound/hda/intel-dsp-config.c | 247 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +- sound/pci/hda/hda_intel.c | 48 ++---- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 +-- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 + 11 files changed, 321 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..c593253c9829 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,8 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +ifneq ($(CONFIG_ACPI),) + snd-intel-dspcfg-objs += intel-nhlt.o +endif +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..3f06377a1808 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,247 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <sound/core.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/dmi.h> +#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); + +#define FLAG_SST (1<<0) +#define FLAG_SOF (1<<1) +#define FLAG_SOF_ONLY_IF_DMIC (1<<16) + +struct config_entry { + u32 flags; + u16 device; + const struct dmi_system_id *dmi_table; +}; + +/* + * configuration table - the order of entries is important! + */ +static const struct config_entry config_table[] = { +/* Cometlake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) + { + /* prefer SST */ + .flags = FLAG_SST, + .device = 0x02c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + { + .flags = FLAG_SOF, + .device = 0x02c8, + }, +#endif +/* Cometlake-H */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) + { + .flags = FLAG_SST, + .device = 0x06c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + { + .flags = FLAG_SOF, + .device = 0x06c8, + }, +#endif +/* Merrifield */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + { + .flags = FLAG_SOF, + .device = 0x119a, + }, +#endif +/* Broxton-T */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x1a98, + }, +#endif +/* Geminilake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { + .flags = FLAG_SOF, + .device = 0x3198, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) + { + .flags = FLAG_SST, + .device = 0x3198, + }, +#endif +/* Icelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x34c8, + }, +#endif +/* Elkhart Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b55, + }, +#endif +/* Appololake (Broxton-P) */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF, + .device = 0x5a98, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Up Squared", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + { + .flags = FLAG_SST, + .device = 0x5a98, + }, +#endif +/* Cannonlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9dc8, + }, +#endif +/* Sunrise Point-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d70, + }, +#endif +/* Kabylake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d71, + }, +#endif +/* Tigerlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa0c8, + }, +#endif +/* Coffelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa348, + }, +#endif +}; + +static const struct config_entry *snd_intel_dsp_find_config + (struct pci_dev *pci, const struct config_entry *table, u32 len) +{ + u16 device; + + device = pci->device; + for (; len > 0; len--, table++) { + if (table->device != device) + continue; + if (table->dmi_table && !dmi_check_system(table->dmi_table)) + continue; + return table; + } + return NULL; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + const struct config_entry *cfg; + + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* find the configuration for the specific device */ + cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table)); + if (!cfg) + return SND_INTEL_DSP_DRIVER_ANY; + + if (cfg->flags & FLAG_SOF) { + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + } else { + return SND_INTEL_DSP_DRIVER_SOF; + } + } + + if (cfg->flags & FLAG_SST) + return SND_INTEL_DSP_DRIVER_SST; + + return SND_INTEL_DSP_DRIVER_LEGACY; +} + +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..7e262678abeb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool no_dsp_driver; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,8 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(no_dsp_driver, bool, 0444); +MODULE_PARM_DESC(no_dsp_driver, "Force this driver to be used and bypass the DSP driver selection"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2020,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2037,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (!no_dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2060,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

3 7

[alsa-devel] [PATCH v4] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 09 Oct '19

09 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems. The examples are taken from the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 34 +++++ sound/hda/Kconfig | 10 +- sound/hda/Makefile | 4 +- sound/hda/intel-dsp-config.c | 249 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +- sound/pci/hda/hda_intel.c | 49 ++---- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 +-- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 + 11 files changed, 321 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..e3903e8f2d97 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,5 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o intel-nhlt.o +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..d9f6d9e872b4 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,249 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <linux/bits.h> +#include <linux/dmi.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/intel-dsp-config.h> +#include <sound/intel-nhlt.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); + +#define FLAG_SST BIT(0) +#define FLAG_SOF BIT(1) +#define FLAG_SOF_ONLY_IF_DMIC BIT(16) + +struct config_entry { + u32 flags; + u16 device; + const struct dmi_system_id *dmi_table; +}; + +/* + * configuration table + * - the order of similar PCI ID entries is important! + * - the first successful match will win + */ +static const struct config_entry config_table[] = { +/* Cometlake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) + { + /* prefer SST */ + .flags = FLAG_SST, + .device = 0x02c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + { + .flags = FLAG_SOF, + .device = 0x02c8, + }, +#endif +/* Cometlake-H */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) + { + .flags = FLAG_SST, + .device = 0x06c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + { + .flags = FLAG_SOF, + .device = 0x06c8, + }, +#endif +/* Merrifield */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + { + .flags = FLAG_SOF, + .device = 0x119a, + }, +#endif +/* Broxton-T */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x1a98, + }, +#endif +/* Geminilake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { + .flags = FLAG_SOF, + .device = 0x3198, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) + { + .flags = FLAG_SST, + .device = 0x3198, + }, +#endif +/* Icelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x34c8, + }, +#endif +/* Elkhart Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b55, + }, +#endif +/* Appololake (Broxton-P) */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF, + .device = 0x5a98, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Up Squared", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + { + .flags = FLAG_SST, + .device = 0x5a98, + }, +#endif +/* Cannonlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9dc8, + }, +#endif +/* Sunrise Point-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d70, + }, +#endif +/* Kabylake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d71, + }, +#endif +/* Tigerlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa0c8, + }, +#endif +/* Coffelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa348, + }, +#endif +}; + +static const struct config_entry *snd_intel_dsp_find_config + (struct pci_dev *pci, const struct config_entry *table, u32 len) +{ + u16 device; + + device = pci->device; + for (; len > 0; len--, table++) { + if (table->device != device) + continue; + if (table->dmi_table && !dmi_check_system(table->dmi_table)) + continue; + return table; + } + return NULL; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + const struct config_entry *cfg; + + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* find the configuration for the specific device */ + cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table)); + if (!cfg) + return SND_INTEL_DSP_DRIVER_ANY; + + if (cfg->flags & FLAG_SOF) { + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + } else { + return SND_INTEL_DSP_DRIVER_SOF; + } + } + + if (cfg->flags & FLAG_SST) + return SND_INTEL_DSP_DRIVER_SST; + + return SND_INTEL_DSP_DRIVER_LEGACY; +} +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..a916963822e0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool dsp_driver = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,9 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(dsp_driver, bool, 0444); +MODULE_PARM_DESC(dsp_driver, "Allow DSP driver selection (bypass this driver) " + "(0=off, 1=on) (default=1)"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2021,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2038,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2061,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: soc-pcm: fixup dpcm_prune_paths() loop continue
by Kuninori Morimoto 09 Oct '19

09 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> dpcm_prune_paths() is checking widget at 2 parts. (A) is for CPU, (B) is for Codec. If we focus to (A) part, continue at (a) is for (1) loop. But, if we focus to (B) part, continue at (b) is for (2) loop, not for (1). This is bug. This patch fixup this issue. static int dpcm_prune_paths(...) { ... (1) for_each_dpcm_be(fe, stream, dpcm) { ... ^ widget = dai_get_widget(...); | (A) if (widget && widget_in_list(...)) | (a) continue; v ^ (2) for_each_rtd_codec_dai(...) { | widget = dai_get_widget(...); (B) | if (widget && widget_in_list(...)) v (b) continue; } ... Fixes: 2e5894d73789 ("ASoC: pcm: Add support for DAI multicodec") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6706b91..5d973ac 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1443,6 +1443,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget *widget; struct snd_soc_dai *dai; int prune = 0; + int do_prune; /* Destroy any old FE <--> BE connections */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1456,13 +1457,16 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* is there a valid CODEC DAI widget for this BE */ + do_prune = 1; for_each_rtd_codec_dai(dpcm->be, i, dai) { widget = dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ if (widget && widget_in_list(list, widget)) - continue; + do_prune = 0; } + if (!do_prune) + continue; dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n", stream ? "capture" : "playback", -- 2.7.4

1 0

[alsa-devel] Is this bug ? dpcm_prune_paths()
by Kuninori Morimoto 09 Oct '19

09 Oct '19

Hi ALSA ML. I'm checking soc-pcm.c, and I noticed strange code at dpcm_prune_paths(). static int dpcm_prune_paths(...) { ... /* Destroy any old FE <--> BE connections */ (1) for_each_dpcm_be(fe, stream, dpcm) { unsigned int i; - /* is there a valid CPU DAI widget for this BE */ | widget = dai_get_widget(dpcm->be->cpu_dai, stream); | (A) /* prune the BE if it's no longer in our active list */ | if (widget && widget_in_list(list, widget)) |(a) continue; - | /* is there a valid CODEC DAI widget for this BE */ |(2) for_each_rtd_codec_dai(dpcm->be, i, dai) { | widget = dai_get_widget(dai, stream); (B) | /* prune the BE if it's no longer in our active list */ | if (widget && widget_in_list(list, widget)) |(b) continue; - } - ... | dpcm->state = ... (C) dpcm->be->dpcm[stream].runtime_update = ... | prune++; - } ... } In my understanding, (A) part is for CPU, and (B) part is for Codec. Guessing from (a), I think it want to skip setup (C) if CPU widget exist and matches to list. If so, I guess (b) is maybe bug ? This continue is for (2) loop, thus, it is totally do nothing now. But maybe it want to be continue for (1) loop ? Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[alsa-devel] [PATCH -next] ASoc: tas2770: Fix build error without GPIOLIB
by YueHaibing 08 Oct '19

08 Oct '19

If GPIOLIB is not set, building fails: sound/soc/codecs/tas2770.c: In function tas2770_reset: sound/soc/codecs/tas2770.c:38:3: error: implicit declaration of function gpiod_set_value_cansleep; did you mean gpio_set_value_cansleep? [-Werror=implicit-function-declaration] gpiod_set_value_cansleep(tas2770->reset_gpio, 0); ^~~~~~~~~~~~~~~~~~~~~~~~ gpio_set_value_cansleep sound/soc/codecs/tas2770.c: In function tas2770_i2c_probe: sound/soc/codecs/tas2770.c:749:24: error: implicit declaration of function devm_gpiod_get_optional; did you mean devm_regulator_get_optional? [-Werror=implicit-function-declaration] tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, ^~~~~~~~~~~~~~~~~~~~~~~ devm_regulator_get_optional sound/soc/codecs/tas2770.c:751:13: error: GPIOD_OUT_HIGH undeclared (first use in this function); did you mean GPIOF_INIT_HIGH? GPIOD_OUT_HIGH); ^~~~~~~~~~~~~~ GPIOF_INIT_HIGH Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac957..d639f17 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1108,6 +1108,7 @@ config SND_SOC_TAS2552 config SND_SOC_TAS2770 tristate "Texas Instruments TAS2770 speaker amplifier" depends on I2C + select GPIOLIB config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" -- 2.7.4

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[alsa-devel] [PATCH v2 0/4] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 08 Oct '19

08 Oct '19

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the following patch: 1. this series add support reset-controller in infra [v5,2/2] clk: reset: Modify reset-controller driver https://patchwork.kernel.org/patch/11060419/ v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on Jiaxin Yu (2): dt-bindings: medaitek: mt8183: add property "resets" && "reset-names" ASoC: mt8183: fix audio playback slowly after playback during bootup yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../bindings/sound/mt8183-afe-pcm.txt | 6 + .../devicetree/bindings/watchdog/mtk-wdt.txt | 9 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 110 +++++++++++++++++- .../reset-controller/mt8183-resets.h | 13 +++ sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 15 +++ 6 files changed, 150 insertions(+), 4 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH 00/44] ASoC: don't use snd_pcm_ops
by Kuninori Morimoto 08 Oct '19

08 Oct '19

Hi Mark Current snd_soc_component_driver is using snd_pcm_ops. But, it is mainly for ALSA, thus, it not good much to ALSA SoC. For example .open case (1)(2), each callback can't know it is called for which component. Thus, it need to use using snd_soc_rtdcom_lookup() (3). --- ALSA SoC --- ... if (component->driver->ops && component->driver->ops->open) (1) return component->driver->ops->open(substream); ... --- driver --- (2) static int xxx_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; (3) struct snd_soc_component *component = snd_soc_rtdcom_lookup(..); ... } It works today, but, will not work in the future if we support multi CPU/Codec/Platform, because 1 rtd might have multiple components which have same driver name. In other words snd_soc_rtdcom_lookup() doesn't work correctly in the future. To solve this issue, each callback needs to be called with component. This patch copies each snd_pcm_ops member under component driver, and having "component" as parameter. --- ALSA SoC --- ... if (component->driver->open) => return component->driver->open(component, substream); ... --- driver --- => static int xxx_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { ... } Kuninori Morimoto (44): ASoC: soc-core: merge snd_pcm_ops member to component driver ASoC: soc-core: add new pcm_construct/pcm_destruct ASoC: soc-core: add snd_soc_pcm_lib_ioctl() ASoC: intel: sst-haswell-pcm: remove snd_pcm_ops ASoC: intel: skl-pcm: remove snd_pcm_ops ASoC: intel: sst-mfld-platform-pcm: remove snd_pcm_ops ASoC: intel: sst-baytrail-pcm: remove snd_pcm_ops ASoC: amd: acp3x-pcm-dma: remove snd_pcm_ops ASoC: amd: acp-pcm-dma: remove snd_pcm_ops ASoC: atmel: atmel-pcm-pdc: remove snd_pcm_ops ASoC: au1x: dbdma2: remove snd_pcm_ops ASoC: au1x: dma: remove snd_pcm_ops ASoC: xtensa: xtfpga-i2s: remove snd_pcm_ops ASoC: xilinx: xlnx_formatter_pcm: remove snd_pcm_ops ASoC: uniphier: aio-dma: remove snd_pcm_ops ASoC: txx9: txx9aclc: remove snd_pcm_ops ASoC: stm: stm32_adfsdm: remove snd_pcm_ops ASoC: sprd: sprd-pcm-dma: remove snd_pcm_ops ASoC: sof: pcm: remove snd_pcm_ops ASoC: sh: rsnd: remove snd_pcm_ops ASoC: sh: fsi: remove snd_pcm_ops ASoC: sh: dma-sh7760: remove snd_pcm_ops ASoC: sh: siu_pcm: remove snd_pcm_ops ASoC: samsung: idma: remove snd_pcm_ops ASoC: qcom: lpass-platform: remove snd_pcm_ops ASoC: qcom: q6routing: remove snd_pcm_ops ASoC: qcom: q6asm-dai: remove snd_pcm_ops ASoC: pxa: mmp-pcm: remove snd_pcm_ops ASoC: pxa: remove snd_pcm_ops ASoC: meson: remove snd_pcm_ops ASoC: mediatek: mtk-btcvsd: remove snd_pcm_ops ASoC: mediatek: remove snd_pcm_ops ASoC: kirkwood: kirkwood-dma: remove snd_pcm_ops ASoC: dwc: dwc-pcm: remove snd_pcm_ops ASoC: rt5514-spi: remove snd_pcm_ops ASoC: bcm: cygnus-pcm: remove snd_pcm_ops ASoC: fsl: imx-pcm-fiq: remove snd_pcm_ops ASoC: fsl: fsl_dma: remove snd_pcm_ops ASoC: fsl: mpc5200_dma: remove snd_pcm_ops ASoC: fsl: fsl_asrc_dma: remove snd_pcm_ops ASoC: fsl: fsl_dma: don't use snd_soc_rtdcom_lookup() ASoC: soc-generic-dmaengine-pcm: remove snd_pcm_ops ASoC: soc-utils: remove snd_pcm_ops ASoC: soc-component: remove snd_pcm_ops from component driver include/sound/pxa2xx-lib.h | 26 +++++- include/sound/soc-component.h | 41 ++++++++- include/sound/soc.h | 5 + sound/arm/pxa2xx-pcm-lib.c | 80 +++++++++++++--- sound/soc/amd/acp-pcm-dma.c | 63 ++++++------- sound/soc/amd/raven/acp3x-pcm-dma.c | 54 +++++------ sound/soc/atmel/atmel-pcm-pdc.c | 48 +++++----- sound/soc/au1x/dbdma2.c | 64 +++++++------ sound/soc/au1x/dma.c | 63 +++++++------ sound/soc/bcm/cygnus-pcm.c | 56 +++++------ sound/soc/codecs/rt5514-spi.c | 41 +++++---- sound/soc/dwc/dwc-pcm.c | 48 +++++----- sound/soc/fsl/fsl_asrc_dma.c | 56 +++++------ sound/soc/fsl/fsl_dma.c | 53 +++++------ sound/soc/fsl/imx-pcm-fiq.c | 56 ++++++----- sound/soc/fsl/mpc5200_dma.c | 51 ++++++----- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 30 +++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 52 +++++------ sound/soc/intel/haswell/sst-haswell-pcm.c | 62 +++++++------ sound/soc/intel/skylake/skl-pcm.c | 67 +++++++++----- sound/soc/kirkwood/kirkwood-dma.c | 50 +++++----- .../soc/mediatek/common/mtk-afe-platform-driver.c | 28 +++--- .../soc/mediatek/common/mtk-afe-platform-driver.h | 10 +- sound/soc/mediatek/common/mtk-btcvsd.c | 76 ++++++--------- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 11 ++- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 11 ++- sound/soc/meson/axg-fifo.c | 56 +++++------ sound/soc/meson/axg-fifo.h | 20 +++- sound/soc/meson/axg-frddr.c | 24 ++++- sound/soc/meson/axg-toddr.c | 24 ++++- sound/soc/pxa/mmp-pcm.c | 62 ++++++++----- sound/soc/pxa/pxa-ssp.c | 14 ++- sound/soc/pxa/pxa2xx-ac97.c | 14 ++- sound/soc/pxa/pxa2xx-i2s.c | 14 ++- sound/soc/pxa/pxa2xx-pcm.c | 14 ++- sound/soc/qcom/lpass-platform.c | 70 +++++++------- sound/soc/qcom/qdsp6/q6asm-dai.c | 71 +++++++------- sound/soc/qcom/qdsp6/q6routing.c | 14 +-- sound/soc/samsung/idma.c | 58 ++++++------ sound/soc/sh/dma-sh7760.c | 46 +++++----- sound/soc/sh/fsi.c | 31 ++++--- sound/soc/sh/rcar/core.c | 23 +++-- sound/soc/sh/siu_pcm.c | 44 +++++---- sound/soc/soc-component.c | 76 +++++++-------- sound/soc/soc-core.c | 12 +++ sound/soc/soc-generic-dmaengine-pcm.c | 102 ++++++++++----------- sound/soc/soc-pcm.c | 11 +-- sound/soc/soc-utils.c | 11 +-- sound/soc/sof/pcm.c | 88 +++++++++--------- sound/soc/sprd/sprd-pcm-dma.c | 74 +++++++-------- sound/soc/stm/stm32_adfsdm.c | 42 +++++---- sound/soc/txx9/txx9aclc.c | 48 +++++----- sound/soc/uniphier/aio-dma.c | 51 ++++++----- sound/soc/xilinx/xlnx_formatter_pcm.c | 54 +++++------ sound/soc/xtensa/xtfpga-i2s.c | 30 +++--- 55 files changed, 1311 insertions(+), 1089 deletions(-) -- 2.7.4

2 89

[alsa-devel] [PATCH v2] firmware: vpd: Add an interface to read VPD value
by Cheng-Yi Chiang 08 Oct '19

08 Oct '19

Add an interface for other driver to query VPD value. This will be used for ASoC machine driver to query calibration data stored in VPD for smart amplifier speaker resistor calibration. The example usage in ASoC machine driver is like: #define DSM_CALIB_KEY "dsm_calib" static int load_calibration_data(struct cml_card_private *ctx) { char *data = NULL; int ret; u32 value_len; /* Read calibration data from VPD. */ ret = vpd_attribute_read(1, DSM_CALIB_KEY, (u8 **)&data, &value_len); /* Parsing of this string...*/ } Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Change from v1 to v2: - Use kmemdup to copy data. - Set value_len according to bin_attr.size. - Check return value of kmemdup. - Rename the function to vpd_attribute_read. - Add docstrings for the function. - Returns -ENOENT when the key is not found. - Use EXPORT_SYMBOL_GPL. Note: The user of this API is in ASoC machine driver cml_rt1011_rt5682. It is pending on the initial machine driver change https://patchwork.kernel.org/patch/11161145/ and the codec driver change to provide API to do calibration. https://patchwork.kernel.org/patch/11179237/ The draft patch of machine driver change which uses this API is at https://chromium-review.googlesource.com/c/chromiumos/third_party/kernel/+/… drivers/firmware/google/vpd.c | 31 ++++++++++++++++++++++ include/linux/firmware/google/google_vpd.h | 18 +++++++++++++ 2 files changed, 49 insertions(+) create mode 100644 include/linux/firmware/google/google_vpd.h diff --git a/drivers/firmware/google/vpd.c b/drivers/firmware/google/vpd.c index db0812263d46..c2be0e756402 100644 --- a/drivers/firmware/google/vpd.c +++ b/drivers/firmware/google/vpd.c @@ -65,6 +65,37 @@ static ssize_t vpd_attrib_read(struct file *filp, struct kobject *kobp, info->bin_attr.size); } +/** + * vpd_attribute_read - Read VPD value for a key. + * @ro: True for RO section. False for RW section. + * @key: A string for key. + * @value: Where to write the VPD value on success. The caller + * must free the memory. + * @value_len: The length of value in bytes. + * + * Returns 0 on success, -ENOENT when the key is not found, and + * -ENOMEM when failed to allocate memory for value. + */ +int vpd_attribute_read(bool ro, const char *key, + u8 **value, u32 *value_len) +{ + struct vpd_attrib_info *info; + struct vpd_section *sec = ro ? &ro_vpd : &rw_vpd; + + list_for_each_entry(info, &sec->attribs, list) { + if (strcmp(info->key, key) == 0) { + *value = kmemdup(info->value, info->bin_attr.size, + GFP_KERNEL); + if (!*value) + return -ENOMEM; + *value_len = info->bin_attr.size; + return 0; + } + } + return -ENOENT; +} +EXPORT_SYMBOL_GPL(vpd_attribute_read); + /* * vpd_section_check_key_name() * diff --git a/include/linux/firmware/google/google_vpd.h b/include/linux/firmware/google/google_vpd.h new file mode 100644 index 000000000000..4364eaa4e1e3 --- /dev/null +++ b/include/linux/firmware/google/google_vpd.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Google VPD interface. + * + * Copyright 2019 Google Inc. + */ + +/* Interface for reading VPD value on Chrome platform. */ + +#ifndef __GOOGLE_VPD_H +#define __GOOGLE_VPD_H + +#include <linux/types.h> + +int vpd_attribute_read(bool ro, const char *key, + u8 **value, u32 *value_len); + +#endif /* __GOOGLE_VPD_H */ -- 2.23.0.581.g78d2f28ef7-goog

2 2

[alsa-devel] [RFC PATCH 00/12] soundwire/SOF: updated interfaces, functional integration
by Pierre-Louis Bossart 08 Oct '19

08 Oct '19

This series builds on the 'soundwire: add Master device support, GreyBus style' RFC'. It provides enhancements to the stream callbacks, a split initialization. Most of the SOF patches were already submitted in an earlier RFC, and feedback on the parameters was taken into account. The main change here are the API changes with a split between ACPI scan, probe, startup steps. Known limits: a) Power management (regular suspend/resume and pm_runtime) is not supported for now as we need to run additional checks on hardware. This will be provided as a separate series. b) during validation checks on CML/ICL, initialization and playback/capture worked fine, but we observed a reproducible system freeze while doing load/unload tests, so likely an initialization missing or a leak to be fixed. Comments and feedback welcome. Pierre-Louis Bossart (7): ASoC: soc-acpi: add link_mask field ASoC: SOF: support alternate list of machines ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: Intel: add build support for SoundWire ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect Rander Wang (5): soundwire: intel: update stream callbacks for hwparams/free stream operations soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback soundwire: intel: free all resources on hw_free() drivers/soundwire/intel.c | 181 ++++++++++++++++++- drivers/soundwire/intel_init.c | 2 +- include/linux/soundwire/sdw_intel.h | 40 ++++- include/sound/soc-acpi.h | 2 + include/sound/sof.h | 3 + include/sound/sof/dai-intel.h | 18 +- sound/soc/sof/intel/Kconfig | 23 +++ sound/soc/sof/intel/hda-loader.c | 9 + sound/soc/sof/intel/hda.c | 261 +++++++++++++++++++++++++++- sound/soc/sof/intel/hda.h | 36 ++++ 10 files changed, 546 insertions(+), 29 deletions(-) -- 2.20.1

2 14

[alsa-devel] [PATCH v2 1/3] ASoC: tas2770: Fix snd_soc_update_bits error handling
by Dan Murphy 07 Oct '19

07 Oct '19

According the documentation for snd_soc_update_bits the API will return a 1 if the update was successful with a value change, a 0 if the update was successful with no value change or a negative if the command just failed. So the value of return in the driver needs to be checked for being less then 0 or the caller may indicate failure when the value actually changed. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v2 - Missed one update_bit return issue. sound/soc/codecs/tas2770.c | 46 +++++++++++++++++++++++++------------- 1 file changed, 30 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index dbbb21fe0548..361d0bba72b3 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -82,7 +82,8 @@ static int tas2770_codec_suspend(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_SHUTDOWN); - if (ret) + + if (ret < 0) return ret; return 0; @@ -96,8 +97,9 @@ static int tas2770_codec_resume(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - if (ret) - return -EINVAL; + + if (ret < 0) + return ret; return 0; } @@ -149,7 +151,10 @@ static int tas2770_dac_event(struct snd_soc_dapm_widget *w, } end: - return ret; + if (ret < 0) + return ret; + + return 0; } static const struct snd_kcontrol_new isense_switch = @@ -199,7 +204,10 @@ static int tas2770_mute(struct snd_soc_dai *dai, int mute) TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) @@ -252,7 +260,10 @@ static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) tas2770->i_sense_slot); end: - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) @@ -344,9 +355,11 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) } end: - if (!ret) - tas2770->sampling_rate = samplerate; - return ret; + if (ret < 0) + return ret; + + tas2770->sampling_rate = samplerate; + return 0; } static int tas2770_hw_params(struct snd_pcm_substream *substream, @@ -401,7 +414,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, TAS2770_TDM_CFG_REG1_RX_MASK, asi_cfg_1); - if (ret) + if (ret < 0) return ret; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -426,7 +439,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, TAS2770_TDM_CFG_REG1_MASK, (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); - if (ret) + if (ret < 0) return ret; value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); @@ -472,12 +485,12 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_30_MASK, (left_slot << TAS2770_TDM_CFG_REG3_30_SHIFT)); - if (ret) + if (ret < 0) return ret; ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_RXS_MASK, (right_slot << TAS2770_TDM_CFG_REG3_RXS_SHIFT)); - if (ret) + if (ret < 0) return ret; switch (slot_width) { @@ -511,10 +524,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } - if (!ret) - tas2770->slot_width = slot_width; + if (ret < 0) + return ret; - return ret; + tas2770->slot_width = slot_width; + return 0; } static struct snd_soc_dai_ops tas2770_dai_ops = { -- 2.22.0.214.g8dca754b1e

2 5

[alsa-devel] [PATCH 0/2] [RFC] ASoC: topology: fix access beyond data
by Guennadi Liakhovetski 07 Oct '19

07 Oct '19

This is likely incomplete, I haven't reviewed the whole of the topology parsing code, and I haven't tested this yet - neither for actually being a problem without these patches nor for fixing them. This is a preview to give everybody a chance to showt out if I misunderstood something and there isn't actually a problem to be fixed. Thanks Guennadi Guennadi Liakhovetski (2): ASoC: topology: protect against accessing beyond loaded topology data ASoC: topology: don't access beyond topology data sound/soc/soc-topology.c | 37 +++++++++++++++++++++++++++++-------- 1 file changed, 29 insertions(+), 8 deletions(-) -- 1.9.3

2 5

[alsa-devel] what's the kernel policy WRT firmware parsing security?
by Guennadi Liakhovetski 07 Oct '19

07 Oct '19

Hi, I decided to have a look at whether the ALSA topology parsing is bullet proof against malformed topology files. It seems not quite to be the case. At least I seem to have found a possibility of crashing the kernel by a malformed topology file. I haven't tested it, so, maybe I'm wrong. In principle, firmware files can only be written by root, and if you have root access to the system, it's anyway doomed. Is this the approach and we aren't really trying to make topology parsing 100% safe, or do we want to fix any such possible parsing issues? Thanks Guennadi

2 2

[alsa-devel] [PATCH 0/1] ALSA: hda: add fallback to polling to hdac_bus_get_response()
by Kai Vehmanen 07 Oct '19

07 Oct '19

Hey all, while debugging issues with some Intel platforms related to display audio codec probe (see https://lists.freedesktop.org/archives/intel-gfx/2019-October/214621.html ), I found a discrepancy in behaviour between snd-hda-intel and SOF, despite using the same snd-hda-codec-hdmi as the codec driver. The specific problem I was debugging appears in a stress test (designed to uncover the above display driver issue) where driver-unload, s3-suspend, resume and driver-reload is done in a loop and repeated for hundreds of iterations. When using SOF, I would get occasional probe fail due to a missing HDA irq. The AZX snd_hda_intel driver nicely survives this test. The explanation seems to be differences in the hdac get_response() implementation. While the specific issue could be solved with other means, the git history shows a number of rare issues with HDA codecs where polling has helped. It would seem best to align the logic with the AZX driver implementation that has seen much more usage over the years. This will benefit SOF and any other users of the HDAC library. Kai Vehmanen (1): ALSA: hda: add fallback to polling to hdac_bus_get_response() sound/hda/hdac_controller.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) -- 2.17.1

2 3

[alsa-devel] [PATCH v2 RESEND] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 07 Oct '19

07 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v1 to v2: - Use audio_lock to protect audio clock. - Fix the patch title. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index aa7efd4da1c8..749d8e4c535b 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -1982,6 +1982,17 @@ static void hdmi_disable_overflow_interrupts(struct dw_hdmi *hdmi) HDMI_IH_MUTE_FC_STAT2); } +static void dw_hdmi_audio_restore(struct dw_hdmi *hdmi) +{ + unsigned long flags; + + spin_lock_irqsave(&hdmi->audio_lock, flags); + + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); + + spin_unlock_irqrestore(&hdmi->audio_lock, flags); +} + static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) { int ret; @@ -2045,7 +2056,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + dw_hdmi_audio_restore(hdmi); } /* not for DVI mode */ -- 2.23.0.444.g18eeb5a265-goog

2 2

[alsa-devel] [PATCH 07/10] ASoC: jz4740: Use of_device_get_match_data()
by Stephen Boyd 07 Oct '19

07 Oct '19

This probe function is only called if the device is backed by a DT node, so switch this call to of_device_get_match_data() to reduce code size and simplify a bit. This also avoids needing to reference a potentially undefined variable because of_device_get_match_data() doesn't need to know anything beyond the struct device to find the match table. Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Paul Cercueil <paul(a)crapouillou.net> 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: Rob Herring <robh+dt(a)kernel.org> Cc: Frank Rowand <frowand.list(a)gmail.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- Please ack or pick for immediate merge so the last patch can be merged. sound/soc/jz4740/jz4740-i2s.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 13408de34055..d2dab4d24b87 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -503,9 +503,8 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) if (!i2s) return -ENOMEM; - match = of_match_device(jz4740_of_matches, &pdev->dev); - if (match) - i2s->version = (enum jz47xx_i2s_version)match->data; + i2s->version = + (enum jz47xx_i2s_version)of_device_get_match_data(&pdev->dev); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); i2s->base = devm_ioremap_resource(&pdev->dev, mem); -- Sent by a computer through tubes

2 1

[alsa-devel] [PATCH] ASoC: core: Fix pcm code debugfs error
by Hans de Goede 07 Oct '19

07 Oct '19

We can have 2 dcpm-s with the same backend and frontend name (capture + playback pair), this causes the following debugfs error on Intel Bay Trail systems: [ 298.969049] debugfs: Directory 'SSP2-Codec' with parent 'Baytrail Audio Port' already present! This commit adds a ":playback" or ":capture" postfix to the debugfs dir name fixing this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/soc-pcm.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..2e723012eb52 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1146,6 +1146,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; unsigned long flags; + char *name; /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1171,9 +1172,15 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, stream ? "<-" : "->", be->dai_link->name); #ifdef CONFIG_DEBUG_FS - dpcm->debugfs_state = debugfs_create_dir(be->dai_link->name, - fe->debugfs_dpcm_root); - debugfs_create_u32("state", 0644, dpcm->debugfs_state, &dpcm->state); + name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name, + stream ? "capture" : "playback"); + if (name) { + dpcm->debugfs_state = debugfs_create_dir(name, + fe->debugfs_dpcm_root); + debugfs_create_u32("state", 0644, dpcm->debugfs_state, + &dpcm->state); + kfree(name); + } #endif return 1; } -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_mqs: Fix error handling in probe
by Dan Carpenter 07 Oct '19

07 Oct '19

There are several problems in the error handling in fsl_mqs_probe(). 1) "ret" isn't initialized on some paths. GCC has a feature which warns about uninitialized variables but the code initializes "ret" to zero at the start of the function so the checking is turned off. 2) "gpr_np" is a pointer so initializing it to zero is confusing and generates a Sparse warning. 3) of_parse_phandle() doesn't return error pointers on error, it returns NULL. 4) If devm_snd_soc_register_component() fails then the function should free the "gpr_np". Fixes: 9e28f6532c61 ("ASoC: fsl_mqs: Add MQS component driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/fsl/fsl_mqs.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c1619a553514..f5f2f659bcbf 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -179,10 +179,10 @@ static const struct regmap_config fsl_mqs_regmap_config = { static int fsl_mqs_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct device_node *gpr_np = 0; + struct device_node *gpr_np = NULL; struct fsl_mqs *mqs_priv; void __iomem *regs; - int ret = 0; + int ret; mqs_priv = devm_kzalloc(&pdev->dev, sizeof(*mqs_priv), GFP_KERNEL); if (!mqs_priv) @@ -199,17 +199,16 @@ static int fsl_mqs_probe(struct platform_device *pdev) if (mqs_priv->use_gpr) { gpr_np = of_parse_phandle(np, "gpr", 0); - if (IS_ERR(gpr_np)) { + if (!gpr_np) { dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); - ret = PTR_ERR(gpr_np); - goto out; + return -EINVAL; } mqs_priv->regmap = syscon_node_to_regmap(gpr_np); if (IS_ERR(mqs_priv->regmap)) { dev_err(&pdev->dev, "failed to get gpr regmap\n"); ret = PTR_ERR(mqs_priv->regmap); - goto out; + goto err_free_gpr_np; } } else { regs = devm_platform_ioremap_resource(pdev, 0); @@ -230,7 +229,7 @@ static int fsl_mqs_probe(struct platform_device *pdev) if (IS_ERR(mqs_priv->ipg)) { dev_err(&pdev->dev, "failed to get the clock: %ld\n", PTR_ERR(mqs_priv->ipg)); - goto out; + return PTR_ERR(mqs_priv->ipg); } } @@ -238,17 +237,21 @@ static int fsl_mqs_probe(struct platform_device *pdev) if (IS_ERR(mqs_priv->mclk)) { dev_err(&pdev->dev, "failed to get the clock: %ld\n", PTR_ERR(mqs_priv->mclk)); - goto out; + ret = PTR_ERR(mqs_priv->mclk); + goto err_free_gpr_np; } dev_set_drvdata(&pdev->dev, mqs_priv); pm_runtime_enable(&pdev->dev); - return devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, &fsl_mqs_dai, 1); -out: - if (!IS_ERR(gpr_np)) - of_node_put(gpr_np); + if (ret) + goto err_free_gpr_np; + return 0; + +err_free_gpr_np: + of_node_put(gpr_np); return ret; } -- 2.20.1

2 1

[alsa-devel] [PATCH -next] ASoC: fsl_mqs: remove set but not used variable 'bclk'
by YueHaibing 07 Oct '19

07 Oct '19

Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/fsl/fsl_mqs.c: In function fsl_mqs_hw_params: sound/soc/fsl/fsl_mqs.c:54:6: warning: variable bclk set but not used [-Wunused-but-set-variable] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/fsl_mqs.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c1619a5..7b9cab3 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -51,10 +51,9 @@ static int fsl_mqs_hw_params(struct snd_pcm_substream *substream, struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); unsigned long mclk_rate; int div, res; - int bclk, lrclk; + int lrclk; mclk_rate = clk_get_rate(mqs_priv->mclk); - bclk = snd_soc_params_to_bclk(params); lrclk = params_rate(params); /* -- 2.7.4

2 1

[alsa-devel] tegra30 tdm audio support
by Ben Dooks 07 Oct '19

07 Oct '19

This adds support for tegra30 tdm audio and fixes a bug in the fifo handling on rx when non-32 bit aligned data is used. v2: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock v3: - cleanup fsync patch - fix rebase issue with tdm patch

3 19

[alsa-devel] [PATCH] ALSA: hdac: clear link output stream mapping
by Pierre-Louis Bossart 07 Oct '19

07 Oct '19

From: Rander Wang <rander.wang(a)linux.intel.com> Fix potential DMA hang upon starting playback on devices in HDA mode on Intel platforms (Gemini Lake/Whiskey Lake/Comet Lake/Ice Lake). It doesn't affect platforms before Gemini Lake or any Intel device in non-HDA mode. The reset value for the LOSDIV register is all output streams valid. Clear this register to invalidate non-existent streams when the bus is powered up. Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/hda_register.h | 3 +++ sound/hda/ext/hdac_ext_controller.c | 5 +++++ 2 files changed, 8 insertions(+) diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 0fd39295b426..057d2a2d0bd0 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -264,6 +264,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 +/* bit0 is reserved, with BIT(1) mapping to stream1 */ +#define ML_LOSIDV_STREAM_MASK 0xFFFE + #define ML_LCTL_SCF_MASK 0xF #define AZX_MLCTL_SPA (0x1 << 16) #define AZX_MLCTL_CPA (0x1 << 23) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 211ca85acd8c..cfab60d88c92 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -270,6 +270,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, ret = snd_hdac_ext_bus_link_power_up(link); + /* + * clear the register to invalidate all the output streams + */ + snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, + ML_LOSIDV_STREAM_MASK, 0); /* * wait for 521usec for codec to report status * HDA spec section 4.3 - Codec Discovery -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360
by Kai-Heng Feng 07 Oct '19

07 Oct '19

Headphone on XPS 9350/9360 produces a background white noise. The The noise level somehow correlates with "Headphone Mic Boost", when it sets to 1 the noise disappears. However, doing this has a side effect, which also decreases the overall headphone volume so I didn't send the patch upstream. The noise was bearable back then, but after commit 717f43d81afc ("ALSA: hda/realtek - Update headset mode for ALC256") the noise exacerbates to a point it starts hurting ears. So let's use the workaround to set "Headphone Mic Boost" to 1 and lock it so it's not touchable by userspace. Fixes: 717f43d81afc ("ALSA: hda/realtek - Update headset mode for ALC256") BugLink: https://bugs.launchpad.net/bugs/1654448 BugLink: https://bugs.launchpad.net/bugs/1845810 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..b5c225a56b98 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5358,6 +5358,17 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1); + snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP); +} + static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5822,6 +5833,7 @@ enum { ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, @@ -6558,6 +6570,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_dell_xps_13_headphone_noise2, + .chained = true, + .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE + }, [ALC293_FIXUP_LENOVO_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -7001,17 +7019,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), - SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), -- 2.17.1

2 1

[alsa-devel] [RESEND TRIVIAL] sound: Fix Kconfig indentation
by Krzysztof Kozlowski 07 Oct '19

07 Oct '19

Adjust indentation from spaces to tab (+optional two spaces) as in coding style with command like: $ sed -e 's/^ /\t/' -i */Kconfig Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Mark Brown <broonie(a)kernel.org> --- sound/core/Kconfig | 28 ++++++++++---------- sound/drivers/Kconfig | 20 +++++++------- sound/firewire/Kconfig | 6 ++--- sound/isa/Kconfig | 18 ++++++------- sound/mips/Kconfig | 12 ++++----- sound/pci/Kconfig | 2 +- sound/soc/cirrus/Kconfig | 14 +++++----- sound/soc/codecs/Kconfig | 32 +++++++++++------------ sound/soc/intel/Kconfig | 2 +- sound/soc/intel/boards/Kconfig | 48 +++++++++++++++++----------------- sound/soc/pxa/Kconfig | 16 ++++++------ sound/soc/qcom/Kconfig | 20 +++++++------- sound/soc/samsung/Kconfig | 8 +++--- sound/soc/sof/imx/Kconfig | 12 ++++----- sound/soc/sof/intel/Kconfig | 22 ++++++++-------- sound/soc/xilinx/Kconfig | 20 +++++++------- sound/soc/zte/Kconfig | 12 ++++----- sound/usb/Kconfig | 32 +++++++++++------------ 18 files changed, 162 insertions(+), 162 deletions(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 4ee79ad6ae22..4044c42d8595 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -72,11 +72,11 @@ config SND_PCM_OSS config SND_PCM_OSS_PLUGINS bool "OSS PCM (digital audio) API - Include plugin system" depends on SND_PCM_OSS - default y + default y help - If you disable this option, the ALSA's OSS PCM API will not - support conversion of channels, formats and rates. It will - behave like most of new OSS/Free drivers in 2.4/2.6 kernels. + If you disable this option, the ALSA's OSS PCM API will not + support conversion of channels, formats and rates. It will + behave like most of new OSS/Free drivers in 2.4/2.6 kernels. config SND_PCM_TIMER bool "PCM timer interface" if EXPERT @@ -128,13 +128,13 @@ config SND_SUPPORT_OLD_API or older). config SND_PROC_FS - bool "Sound Proc FS Support" if EXPERT - depends on PROC_FS - default y - help - Say 'N' to disable Sound proc FS, which may reduce code size about - 9KB on x86_64 platform. - If unsure say Y. + bool "Sound Proc FS Support" if EXPERT + depends on PROC_FS + default y + help + Say 'N' to disable Sound proc FS, which may reduce code size about + 9KB on x86_64 platform. + If unsure say Y. config SND_VERBOSE_PROCFS bool "Verbose procfs contents" @@ -142,8 +142,8 @@ config SND_VERBOSE_PROCFS default y help Say Y here to include code for verbose procfs contents (provides - useful information to developers when a problem occurs). On the - other side, it makes the ALSA subsystem larger. + useful information to developers when a problem occurs). On the + other side, it makes the ALSA subsystem larger. config SND_VERBOSE_PRINTK bool "Verbose printk" @@ -164,7 +164,7 @@ config SND_DEBUG_VERBOSE depends on SND_DEBUG help Say Y here to enable extra-verbose debugging messages. - + Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages. So, say Y only if you are ready to be annoyed. diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 09932cc98e9d..15d6d46acf9c 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_MPU401_UART - tristate - select SND_RAWMIDI + tristate + select SND_RAWMIDI config SND_OPL3_LIB tristate @@ -90,16 +90,16 @@ config SND_DUMMY will be called snd-dummy. config SND_ALOOP - tristate "Generic loopback driver (PCM)" - select SND_PCM - help - Say 'Y' or 'M' to include support for the PCM loopback device. + tristate "Generic loopback driver (PCM)" + select SND_PCM + help + Say 'Y' or 'M' to include support for the PCM loopback device. This module returns played samples back to the user space using the standard ALSA PCM device. The devices are routed 0->1 and - 1->0, where first number is the playback PCM device and second + 1->0, where first number is the playback PCM device and second number is the capture device. Module creates two PCM devices and configured number of substreams (see the pcm_substreams module - parameter). + parameter). The loopback device allows time sychronization with an external timing source using the time shift universal control (+-20% @@ -142,12 +142,12 @@ config SND_MTS64 select SND_RAWMIDI help The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with - additional SMPTE Timecode capabilities for the parallel port. + additional SMPTE Timecode capabilities for the parallel port. Say 'Y' to include support for this device. To compile this driver as a module, chose 'M' here: the module - will be called snd-mts64. + will be called snd-mts64. config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index b0a904cdb932..995c2cefc222 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -77,7 +77,7 @@ config SND_BEBOB tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" select SND_FIREWIRE_LIB select SND_HWDEP - help + help Say Y here to include support for FireWire devices based on BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware: * Edirol FA-66/FA-101 @@ -111,8 +111,8 @@ config SND_BEBOB * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro - To compile this driver as a module, choose M here: the module - will be called snd-bebob. + To compile this driver as a module, choose M here: the module + will be called snd-bebob. config SND_FIREWIRE_DIGI00X tristate "Digidesign Digi 002/003 family support" diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b690ed937cbe..6ffa48dd5983 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -2,22 +2,22 @@ # ALSA ISA drivers config SND_WSS_LIB - tristate - select SND_PCM + tristate + select SND_PCM select SND_TIMER config SND_SB_COMMON - tristate + tristate config SND_SB8_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON config SND_SB16_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON menuconfig SND_ISA bool "ISA sound devices" diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 8a33402fd415..b497b803c834 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -14,15 +14,15 @@ config SND_SGI_O2 tristate "SGI O2 Audio" depends on SGI_IP32 select SND_PCM - help - Sound support for the SGI O2 Workstation. + help + Sound support for the SGI O2 Workstation. config SND_SGI_HAL2 - tristate "SGI HAL2 Audio" - depends on SGI_HAS_HAL2 + tristate "SGI HAL2 Audio" + depends on SGI_HAS_HAL2 select SND_PCM - help - Sound support for the SGI Indy and Indigo2 Workstation. + help + Sound support for the SGI Indy and Indigo2 Workstation. endif # SND_MIPS diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7630f808d087..93bc9bef7641 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -217,7 +217,7 @@ config SND_CMIPCI will be called snd-cmipci. config SND_OXYGEN_LIB - tristate + tristate config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 2333efac758a..8039a8febefa 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -33,13 +33,13 @@ config SND_EP93XX_SOC_AC97 select SND_SOC_AC97_BUS config SND_EP93XX_SOC_SNAPPERCL15 - tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C - select SND_EP93XX_SOC_I2S - select SND_SOC_TLV320AIC23_I2C - help - Say Y or M here if you want to add support for I2S audio on the - Bluewater Systems Snapper CL15 module. + tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C + select SND_EP93XX_SOC_I2S + select SND_SOC_TLV320AIC23_I2C + help + Say Y or M here if you want to add support for I2S audio on the + Bluewater Systems Snapper CL15 module. config SND_EP93XX_SOC_SIMONE tristate "SoC Audio support for Simplemachines Sim.One board" diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac95785493..ed8ba18d93d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -258,16 +258,16 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - help - Normally ASoC codec drivers are only built if a machine driver which - uses them is also built since they are only usable with a machine - driver. Selecting this option will allow these drivers to be built - without an explicit machine driver for test and development purposes. + help + Normally ASoC codec drivers are only built if a machine driver which + uses them is also built since they are only usable with a machine + driver. Selecting this option will allow these drivers to be built + without an explicit machine driver for test and development purposes. Support for the bus types used to access the codecs to be built must be selected separately. - If unsure select "N". + If unsure select "N". config SND_SOC_88PM860X tristate @@ -571,8 +571,8 @@ config SND_SOC_CS42XX8_I2C # Cirrus Logic CS43130 HiFi DAC config SND_SOC_CS43130 - tristate "Cirrus Logic CS43130 CODEC" - depends on I2C + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C config SND_SOC_CS4341 tristate "Cirrus Logic CS4341 CODEC" @@ -644,19 +644,19 @@ config SND_SOC_L3 tristate config SND_SOC_DA7210 - tristate + tristate config SND_SOC_DA7213 - tristate + tristate config SND_SOC_DA7218 tristate config SND_SOC_DA7219 - tristate + tristate config SND_SOC_DA732X - tristate + tristate config SND_SOC_DA9055 tristate @@ -718,7 +718,7 @@ config SND_SOC_INNO_RK3036 select REGMAP_MMIO config SND_SOC_ISABELLE - tristate + tristate config SND_SOC_LM49453 tristate @@ -989,7 +989,7 @@ config SND_SOC_RT5640 tristate config SND_SOC_RT5645 - tristate + tristate config SND_SOC_RT5651 tristate @@ -1225,7 +1225,7 @@ config SND_SOC_UDA134X tristate config SND_SOC_UDA1380 - tristate + tristate depends on I2C config SND_SOC_WCD9335 @@ -1353,7 +1353,7 @@ config SND_SOC_WM8904 depends on I2C config SND_SOC_WM8940 - tristate + tristate config SND_SOC_WM8955 tristate diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..597516c0c192 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -113,7 +113,7 @@ config SND_SOC_INTEL_SKYLAKE select SND_SOC_INTEL_CNL select SND_SOC_INTEL_CFL help - This is a backwards-compatible option to select all devices + This is a backwards-compatible option to select all devices supported by the Intel SST/Skylake driver. This option is no longer recommended and will be deprecated when the SOF driver is introduced. Distributions should explicitly diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7ab4a5f..882ff36a7c9c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -3,13 +3,13 @@ menuconfig SND_SOC_INTEL_MACH bool "Intel Machine drivers" depends on SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL help - Intel ASoC Machine Drivers. If you have a Intel machine that - has an audio controller with a DSP and I2S or DMIC port, then - enable this option by saying Y + Intel ASoC Machine Drivers. If you have a Intel machine that + has an audio controller with a DSP and I2S or DMIC port, then + enable this option by saying Y - Note that the answer to this question doesn't directly affect the - kernel: saying N will just cause the configurator to skip all - the questions about Intel ASoC machine drivers. + Note that the answer to this question doesn't directly affect the + kernel: saying N will just cause the configurator to skip all + the questions about Intel ASoC machine drivers. if SND_SOC_INTEL_MACH @@ -114,11 +114,11 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5670 - help - This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell - platforms with RT5672 audio codec. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" @@ -311,20 +311,20 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH If unsure select "N". config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH - tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" + tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SPI - select SND_SOC_RT5663 - select SND_SOC_RT5514 - select SND_SOC_RT5514_SPI - select SND_SOC_MAX98927 - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for RT5663 + RT5514 + MAX98927. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + depends on SPI + select SND_SOC_RT5663 + select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI + select SND_SOC_MAX98927 + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for RT5663 + RT5514 + MAX98927. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" @@ -393,7 +393,7 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH help This adds support for ASoC machine driver for Intel platforms SKL/KBL/BXT/APL with iDisp, HDA audio codecs. - Say Y or m if you have such a device. This is a recommended option. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 213d4dab0346..295cfffa4646 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -190,14 +190,14 @@ config SND_PXA2XX_SOC_MAGICIAN HTC Magician. config SND_PXA2XX_SOC_MIOA701 - tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 depends on AC97_BUS=n - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9713 - help - Say Y if you want to add support for SoC audio on the - MIO A701. + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" @@ -205,7 +205,7 @@ config SND_PXA2XX_SOC_IMOTE2 select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help - Say Y if you want to add support for SoC audio on the + Say Y if you want to add support for SoC audio on the IMote 2. config SND_MMP_SOC_BROWNSTONE diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 60086858e920..6530d2462a9e 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -3,8 +3,8 @@ config SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST help - Say Y or M if you want to add support to use audio devices - in Qualcomm Technologies SOC-based platforms. + Say Y or M if you want to add support to use audio devices + in Qualcomm Technologies SOC-based platforms. config SND_SOC_LPASS_CPU tristate @@ -30,17 +30,17 @@ config SND_SOC_STORM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help - Say Y or M if you want add support for SoC audio on the - Qualcomm Technologies IPQ806X-based Storm board. + Say Y or M if you want add support for SoC audio on the + Qualcomm Technologies IPQ806X-based Storm board. config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 help - Support for Qualcomm Technologies LPASS audio block in - APQ8016 SOC-based systems. - Say Y if you want to use audio devices on MI2S. + Support for Qualcomm Technologies LPASS audio block in + APQ8016 SOC-based systems. + Say Y if you want to use audio devices on MI2S. config SND_SOC_QCOM_COMMON tristate @@ -93,9 +93,9 @@ config SND_SOC_MSM8996 select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON help - Support for Qualcomm Technologies LPASS audio block in - APQ8096 SoC-based systems. - Say Y if you want to use audio device on this SoCs + Support for Qualcomm Technologies LPASS audio block in + APQ8096 SoC-based systems. + Say Y if you want to use audio device on this SoCs config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 638983123d8f..9304177de78a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -195,10 +195,10 @@ config SND_SOC_ODROID Say Y here to enable audio support for the Odroid XU3/XU4. config SND_SOC_ARNDALE_RT5631_ALC5631 - tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on I2C - select SND_SAMSUNG_I2S - select SND_SOC_RT5631 + tristate "Audio support for RT5631(ALC5631) on Arndale Board" + depends on I2C + select SND_SAMSUNG_I2S + select SND_SOC_RT5631 config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750..6315fba8ce71 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -5,9 +5,9 @@ config SND_SOC_SOF_IMX_TOPLEVEL depends on ARM64|| COMPILE_TEST depends on SND_SOC_SOF_OF help - This adds support for Sound Open Firmware for NXP i.MX platforms. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX platforms. + Say Y if you have such a device. + If unsure select "N". if SND_SOC_SOF_IMX_TOPLEVEL @@ -16,8 +16,8 @@ config SND_SOC_SOF_IMX8 depends on IMX_SCU depends on IMX_DSP help - This adds support for Sound Open Firmware for NXP i.MX8 platforms - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX8 platforms + Say Y if you have such a device. + If unsure select "N". endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index d62f51d33be1..56f16e4543c0 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -36,7 +36,7 @@ config SND_SOC_SOF_INTEL_PCI config SND_SOC_SOF_INTEL_HIFI_EP_IPC tristate help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_INTEL_ATOM_HIFI_EP @@ -217,31 +217,31 @@ config SND_SOC_SOF_COMETLAKE_H_SUPPORT config SND_SOC_SOF_TIGERLAKE_SUPPORT bool "SOF support for Tigerlake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the Tigerlake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the Tigerlake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_TIGERLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_ELKHARTLAKE_SUPPORT bool "SOF support for ElkhartLake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the ElkhartLake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the ElkhartLake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_ELKHARTLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_HDA_COMMON diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 69973179ef15..1d3586b68db7 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,15 +9,15 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" - help - Select this option to enable Xilinx audio formatter - support. This provides DMA platform device support for - audio functionality. + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" - help - Select this option to enable Xilinx SPDIF Audio. - This provides playback and capture of SPDIF audio in - AES format. + tristate "Audio support for the the Xilinx SPDIF" + help + Select this option to enable Xilinx SPDIF Audio. + This provides playback and capture of SPDIF audio in + AES format. diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig index a7842e4b791c..a23d4f13ca19 100644 --- a/sound/soc/zte/Kconfig +++ b/sound/soc/zte/Kconfig @@ -18,9 +18,9 @@ config ZX_I2S ZTE ZX I2S interface config ZX_TDM - tristate "ZTE ZX TDM Driver Support" - depends on COMMON_CLK - select SND_SOC_GENERIC_DMAENGINE_PCM - help - Say Y or M if you want to add support for codecs attached to the - ZTE ZX TDM interface + tristate "ZTE ZX TDM Driver Support" + depends on COMMON_CLK + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for codecs attached to the + ZTE ZX TDM interface diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e2c53a0841da..059242f15d75 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -107,24 +107,24 @@ config SND_USB_US122L will be called snd-usb-us122l. config SND_USB_6FIRE - tristate "TerraTec DMX 6Fire USB" - select FW_LOADER - select BITREVERSE - select SND_RAWMIDI - select SND_PCM - select SND_VMASTER - help - Say Y here to include support for TerraTec 6fire DMX USB interface. - - You will need firmware files in order to be able to use the device - after it has been coldstarted. An install script for the firmware - and further help can be found at - http://sixfireusb.sourceforge.net + tristate "TerraTec DMX 6Fire USB" + select FW_LOADER + select BITREVERSE + select SND_RAWMIDI + select SND_PCM + select SND_VMASTER + help + Say Y here to include support for TerraTec 6fire DMX USB interface. + + You will need firmware files in order to be able to use the device + after it has been coldstarted. An install script for the firmware + and further help can be found at + http://sixfireusb.sourceforge.net config SND_USB_HIFACE - tristate "M2Tech hiFace USB-SPDIF driver" - select SND_PCM - help + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help Select this option to include support for M2Tech hiFace USB-SPDIF interface. -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 06 Oct '19

06 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems like proposed in the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 34 ++++++++++ sound/hda/Kconfig | 10 ++- sound/hda/Makefile | 7 ++- sound/hda/intel-dsp-config.c | 105 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +--- sound/pci/hda/hda_intel.c | 48 +++++--------- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 ++---- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 ++ 11 files changed, 179 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..c593253c9829 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,8 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +ifneq ($(CONFIG_ACPI),) + snd-intel-dspcfg-objs += intel-nhlt.o +endif +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..f331e2216fe0 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,105 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <sound/core.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=noDSP, 2=legacy, 3=SST, 4=SOF)"); + +static const u16 sof_skl_table[] = { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + 0x02c8, /* Cometlake-LP */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + 0x06c8, /* Cometlake-H */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + 0x3198, /* Geminilake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + 0x9dc8, /* Cannonlake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + 0xa348, /* Coffelake */ +#endif +}; + +static int snd_intel_dsp_check_device(u16 device, const u16 *table, u32 len) +{ + for (; len > 0; len--, table++) { + if (*table == device) + return 1; + } + return 0; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + if (snd_intel_dsp_check_device(pci->device, sof_skl_table, ARRAY_SIZE(sof_skl_table))) { + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* Prefer SST driver for Broxton-P (Appololake) */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) || IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + if (pci->device == 0x5a98) +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + return SND_INTEL_DSP_DRIVER_SST; +#else + return SND_INTEL_DSP_DRIVER_SOF; +#endif +#endif + + /* DMIC check for Skylake+ */ + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + + return SND_INTEL_DSP_DRIVER_ANY; +} + +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..7e262678abeb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool no_dsp_driver; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,8 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(no_dsp_driver, bool, 0444); +MODULE_PARM_DESC(no_dsp_driver, "Force this driver to be used and bypass the DSP driver selection"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2020,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2037,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (!no_dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2060,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

2 2

[alsa-devel] [PATCH -next] ASoC: tas2770: fix platform_no_drv_owner.cocci warnings
by YueHaibing 06 Oct '19

06 Oct '19

Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/tas2770.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 9da88ccb1d51..dbbb21fe0548 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -793,7 +793,6 @@ MODULE_DEVICE_TABLE(of, tas2770_of_match); static struct i2c_driver tas2770_i2c_driver = { .driver = { .name = "tas2770", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tas2770_of_match), }, .probe = tas2770_i2c_probe,

1 0

[alsa-devel] [PATCH 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 05 Oct '19

05 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Tzung-Bi Shih (10): platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX platform/chrome: cros_ec: add common commands for EC codec ASoC: cros_ec_codec: read max DMIC gain from EC codec ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1125 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1292 insertions(+), 326 deletions(-) -- 2.23.0.581.g78d2f28ef7-goog

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[alsa-devel] alsa compliance test: H340 (USB audio) playback / capture rate asymmetry bug?
by Hin-Tak Leung 05 Oct '19

05 Oct '19

I am using Takashi Iwai's tree grafted onto mainline as DKMS ( as in https://github.com/HinTak/sound-usb-dkms ). Running the alsa compliance test commit f6167eb77d038b5b7a0d39645e7f2ae7fee6fdc0 (origin/stabilize-12464.B, origin/release-R78-12499.B) Capture all passed, but play back failed a couple, regarding the sample rate. It is a small head set with stereo head phone and a mic. ID 046d:0a38 Logitech, Inc. Headset H340 The question is, for such a cheap headset, why would the playback rate and the capture rate to be different? For many applications/ usages, with a device that's most capture and playback capable, we would like the rates to agree - both pass, or both fail in the same direction? Is this a hardware or software issue? Or, somebody suggested, I haven't looked, issue with the alsa compliance test itself, possibly regarding frame counts of usb devices? detail below. 5 passed, 0 failed Device Information Name: hw:CARD=H340 Stream: CAPTURE Format: ['S16_LE'] Channels range: [2, 2] Rate: [44100] Period_size range: [45, 131072] Buffer_size range: [90, 262144] Test Params Set channels 2: pass Set format S16_LE: pass Set rate 44100: pass Test Rates Set rate 44100: pass Test All Pairs Set channels 2, format S16_LE, rate 44100: pass 3 passed, 2 failed Device Information Name: hw:CARD=H340 Stream: PLAYBACK Format: ['S16_LE'] Channels range: [2, 2] Rate: [44100] Period_size range: [45, 131072] Buffer_size range: [90, 262144] Test Params Set channels 2: pass Set format S16_LE: pass Set rate 44100: pass Test Rates Set rate 44100: fail - Expected rate is 44100.000000, measure 44092.918362, difference 7.081638 > threshold 4.410000 Test All Pairs Set channels 2, format S16_LE, rate 44100: fail - Expected rate is 44100.000000, measure 44093.049192, difference 6.950808 > threshold 4.410000

4 7

[alsa-devel] [RFC PATCH 0/2] Alter the trigger order for FE/BE DAI's based on command
by Ranjani Sridharan 05 Oct '19

05 Oct '19

Currently, the trigger orders SND_SOC_DPCM_TRIGGER_PRE/POST determine the order in which FE DAI and BE DAI are triggered. In the case of SND_SOC_DPCM_TRIGGER_PRE, the FE DAI is triggered before the BE DAI and in the case of SND_SOC_DPCM_TRIGGER_POST, the BE DAI is triggered before the FE DAI. And this order remains the same irrespective of the trigger command. In the case of the SOF driver, during playback, the FW expects the BE DAI to be triggered before the FE DAI during the START trigger. The BE DAI trigger handles the starting of Link DMA and so it must be started before the FE DAI is started to prevent xruns during pause/release. This can be addressed by setting the trigger order for the FE dai link to SND_SOC_DPCM_TRIGGER_POST. But during the STOP trigger, the FW expects the FE DAI to be triggered before the BE DAI. Retaining the same order during the START and STOP commands, results in FW error as the DAI component in the FW is still active. The issue can be fixed by mirroring the trigger order of FE and BE DAI's during the START and STOP trigger. So, with the trigger order set to SND_SOC_DPCM_TRIGGER_PRE, the FE DAI will be trigger first during SNDRV_PCM_TRIGGER_START/STOP/RESUME and the BE DAI will be triggered first during the STOP/SUSPEND/PAUSE commands. Conversely, with the trigger order set to SND_SOC_DPCM_TRIGGER_POST, the BE DAI will be triggered first during the SNDRV_PCM_TRIGGER_START/STOP/RESUME commands and the FE DAI will be triggered first during the SNDRV_PCM_TRIGGER_STOP/SUSPEND/PAUSE commands. My first thought was to use a BESPOKE trigger for the SOF driver but looking more into the implementation of the bespoke trigger in soc_pcm_bespoke_trigger() didnt indicate it had much to do with the ordering of the FE/BE DAI's but rather just use the bespoke trigger callbacks in the DAI driver. More details on the SOF issue can be found in: Github Issue: https://github.com/thesofproject/linux/issues/1160 Ranjani Sridharan (2): ASoC: pcm: update FE/BE trigger order based on the command ASoC: SOF: topology: set trigger order for FE DAI link sound/soc/soc-pcm.c | 99 +++++++++++++++++++++++++++++----------- sound/soc/sof/topology.c | 4 ++ 2 files changed, 76 insertions(+), 27 deletions(-) -- 2.17.1

3 5

[alsa-devel] [PATCH] ASoC: tas2770: Fix snd_soc_update_bits error handling
by Dan Murphy 04 Oct '19

04 Oct '19

According the documentation for snd_soc_update_bits the API will return a 1 if the update was successful with a value change, a 0 if the update was successful with no value change or a negative if the command just failed. So the value of return in the driver needs to be checked for being less then 0 or the caller may indicate failure when the value actually changed. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- arch/arm/configs/omap2plus_defconfig | 1 + sound/soc/codecs/tas2770.c | 44 ++++++++++++++++++---------- 2 files changed, 30 insertions(+), 15 deletions(-) diff --git a/arch/arm/configs/omap2plus_defconfig b/arch/arm/configs/omap2plus_defconfig index 64eb896907bf..8a57185112a2 100644 --- a/arch/arm/configs/omap2plus_defconfig +++ b/arch/arm/configs/omap2plus_defconfig @@ -395,6 +395,7 @@ CONFIG_SND_SOC_OMAP_ABE_TWL6040=m CONFIG_SND_SOC_OMAP_HDMI=m CONFIG_SND_SOC_CPCAP=m CONFIG_SND_SOC_TLV320AIC23_I2C=m +CONFIG_SND_SOC_TAS2770=m CONFIG_SND_SIMPLE_CARD=m CONFIG_SND_AUDIO_GRAPH_CARD=m CONFIG_HID_GENERIC=m diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index dbbb21fe0548..774c8f2833c8 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -82,7 +82,8 @@ static int tas2770_codec_suspend(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_SHUTDOWN); - if (ret) + + if (ret < 0) return ret; return 0; @@ -96,8 +97,9 @@ static int tas2770_codec_resume(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - if (ret) - return -EINVAL; + + if (ret < 0) + return ret; return 0; } @@ -149,7 +151,10 @@ static int tas2770_dac_event(struct snd_soc_dapm_widget *w, } end: - return ret; + if (ret < 0) + return ret; + + return 0; } static const struct snd_kcontrol_new isense_switch = @@ -199,7 +204,10 @@ static int tas2770_mute(struct snd_soc_dai *dai, int mute) TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) @@ -252,7 +260,10 @@ static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) tas2770->i_sense_slot); end: - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) @@ -344,9 +355,11 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) } end: - if (!ret) - tas2770->sampling_rate = samplerate; - return ret; + if (ret < 0) + return ret; + + tas2770->sampling_rate = samplerate; + return 0; } static int tas2770_hw_params(struct snd_pcm_substream *substream, @@ -426,7 +439,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, TAS2770_TDM_CFG_REG1_MASK, (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); - if (ret) + if (ret < 0) return ret; value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); @@ -472,12 +485,12 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_30_MASK, (left_slot << TAS2770_TDM_CFG_REG3_30_SHIFT)); - if (ret) + if (ret < 0) return ret; ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_RXS_MASK, (right_slot << TAS2770_TDM_CFG_REG3_RXS_SHIFT)); - if (ret) + if (ret < 0) return ret; switch (slot_width) { @@ -511,10 +524,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } - if (!ret) - tas2770->slot_width = slot_width; + if (ret < 0) + return ret; - return ret; + tas2770->slot_width = slot_width; + return 0; } static struct snd_soc_dai_ops tas2770_dai_ops = { -- 2.22.0.214.g8dca754b1e

2 2

[alsa-devel] [PATCH] ASoC: wm_adsp: Don't generate kcontrols without READ flags
by Charles Keepax 04 Oct '19

04 Oct '19

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> User space always expects to be able to read ALSA controls, so ensure no kcontrols are generated without an appropriate READ flag. In the case of a read of such a control zeros will be returned. Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 85396d920e0a9..9b8bb7bbe945d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1259,8 +1259,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) } if (in) { - if (in & WMFW_CTL_FLAG_READABLE) - out |= rd; + out |= rd; if (in & WMFW_CTL_FLAG_WRITEABLE) out |= wr; if (in & WMFW_CTL_FLAG_VOLATILE) -- 2.11.0

2 1

[alsa-devel] [PATCH v4 1/3] ASoC: samsung: arndale: Simplify DAI link initialization
by Sylwester Nawrocki 04 Oct '19

04 Oct '19

There is only one DAI link so we can drop an unnecessary loop statement. Use card->dai_link in place of direct static arndale_rt5631_dai[] array dereference as a prerequisite for adding support for other CODECs. Unnecessary assignment of dai_link->codecs->name to NULL is removed. Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Based on for-next branch from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound Changes since v2: - none. Changes since v1: - rebased due to reordering patches in the series --- sound/soc/samsung/arndale_rt5631.c | 42 +++++++++++++----------------- 1 file changed, 18 insertions(+), 24 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index fd8c6642fb0d..004c84fafad9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -88,36 +88,30 @@ static void arndale_put_of_nodes(struct snd_soc_card *card) static int arndale_audio_probe(struct platform_device *pdev) { - int n, ret; struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &arndale_rt5631; + struct snd_soc_dai_link *dai_link; + int ret; card->dev = &pdev->dev; + dai_link = card->dai_link; + + dai_link->cpus->of_node = of_parse_phandle(np, "samsung,audio-cpu", 0); + if (!dai_link->cpus->of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-cpu' missing or invalid\n"); + return -EINVAL; + } - for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) { - if (!arndale_rt5631_dai[n].cpus->dai_name) { - arndale_rt5631_dai[n].cpus->of_node = of_parse_phandle(np, - "samsung,audio-cpu", n); - - if (!arndale_rt5631_dai[n].cpus->of_node) { - dev_err(&pdev->dev, - "Property 'samsung,audio-cpu' missing or invalid\n"); - return -EINVAL; - } - } - if (!arndale_rt5631_dai[n].platforms->name) - arndale_rt5631_dai[n].platforms->of_node = - arndale_rt5631_dai[n].cpus->of_node; - - arndale_rt5631_dai[n].codecs->name = NULL; - arndale_rt5631_dai[n].codecs->of_node = of_parse_phandle(np, - "samsung,audio-codec", n); - if (!arndale_rt5631_dai[0].codecs->of_node) { - dev_err(&pdev->dev, + if (!dai_link->platforms->name) + dai_link->platforms->of_node = dai_link->cpus->of_node; + + dai_link->codecs->of_node = of_parse_phandle(np, "samsung,audio-codec", 0); + if (!dai_link->codecs->of_node) { + dev_err(&pdev->dev, "Property 'samsung,audio-codec' missing or invalid\n"); - ret = -EINVAL; - goto err_put_of_nodes; - } + ret = -EINVAL; + goto err_put_of_nodes; } ret = devm_snd_soc_register_card(card->dev, card); -- 2.17.1

2 5

[alsa-devel] [asoc:for-5.5 32/32] sound/soc/codecs/tas2770.c:796:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 04 Oct '19

04 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: 1a476abc723e644248dba975b71122fcf878703b [32/32] tas2770: add tas2770 smart PA kernel driver If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2770.c:796:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [PATCH 0/6][resend] ASoC: soc-core cleanup - step 3
by Kuninori Morimoto 04 Oct '19

04 Oct '19

Hi Mark Some soc-core cleanup patch step3 are accepted, but not all. These are remains. Kuninori Morimoto (6): ASoC: soc-core: use devm_kzalloc() for rtd ASoC: soc-core: remove soc_remove_dai_links() ASoC: soc-core: add soc_setup_card_name() ASoC: soc-core: use devm_xxx for component related resource ASoC: soc-core: setup card->rtd_num at snd_soc_instantiate_card() ASoC: soc-ops: use snd_soc_card_get_kcontrol() at snd_soc_limit_volume() sound/soc/soc-core.c | 109 +++++++++++++++++++++++++++------------------------ sound/soc/soc-ops.c | 11 +----- 2 files changed, 59 insertions(+), 61 deletions(-) -- 2.7.4 Thank you for your help !! Best regards --- Kuninori Morimoto

3 18

[alsa-devel] [PATCH] ASoc: rockchip: i2s: Fix RPM imbalance
by Robin Murphy 04 Oct '19

04 Oct '19

If rockchip_pcm_platform_register() fails, e.g. upon deferring to wait for an absent DMA channel, we return without disabling RPM, which makes subsequent re-probe attempts scream with errors about the unbalanced enable. Don't do that. Fixes: ebb75c0bdba2 ("ASoC: rockchip: i2s: Adjust devm usage") Signed-off-by: Robin Murphy <robin.murphy(a)arm.com> --- sound/soc/rockchip/rockchip_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index af2d5a6124c8..61c984f10d8e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -677,7 +677,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) ret = rockchip_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - return ret; + goto err_suspend; } return 0; -- 2.21.0.dirty

2 1

[alsa-devel] [PATCH] ASoC: rsnd: Document r8a774b1 bindings
by Biju Das 04 Oct '19

04 Oct '19

Document SoC specific bindings for RZ/G2N (r8a774b1) SoC. Signed-off-by: Biju Das <biju.das(a)bp.renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 5c52182..797fd03 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -268,6 +268,7 @@ Required properties: - "renesas,rcar_sound-r8a7745" (RZ/G1E) - "renesas,rcar_sound-r8a77470" (RZ/G1C) - "renesas,rcar_sound-r8a774a1" (RZ/G2M) + - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - "renesas,rcar_sound-r8a7778" (R-Car M1A) - "renesas,rcar_sound-r8a7779" (R-Car H1) -- 2.7.4

2 1

Re: [alsa-devel] Custom Kernel driver to open and use another ALSA PCM
by Kai Vehmanen 04 Oct '19

04 Oct '19

Hi, (routing back to alsa-devel, Ed's alsa-devel subscription is still pending so he can't sent to the list, forwarding with his permission) On Fri, 4 Oct 2019, ed nwave wrote: > The example used of a voice changing app was just an illustration of how > I need to connect "my driver" directly to a sound card driver that's > all. I understand the user-space comment entirely. The actual > objective is one common driver ( my PCM driver ) being used by 4 music > applications where I can dynamically switch the output PCM without > restarting the applications or the applications even knowing the output > cards not real. so far this is pretty much directly from original design goals of JACK. But ok, then you mention this: > I must have direct access to audio hardware i.e. hw:1,0 > for applications like RoonReady/ RAAT. So Pulseaudio's out. I require > the ability to Now this can be a problem of course if you cannot modify the apps. The you would have to route this via alsa-lib plugins. > follow ( like and MPD playlist ). As I understand Jack is set for a > certain rate and format at the start when connecting and does not update > rates automatically if the rate of the songs change; can jack connect > directly to hw and handle DSD rates?. This is all part of a headless I've been out of the loop for some years now, but no, I don't think JACK supports these. But you are anyways customizing quite a lot of SW, so you could e.g. consider extending JACK and/or some other audio server for this purpose, or just write a new one. > Other requirements are my applications must have no plugins or dmix and > connect directly to hardware i.e. hw:1,0 - due to rate demands, no > remixing. Another requirement is the latency required by one company to > attain multi-room streaming being about 1ms max, this is for > certification. Finally there's the need to change bit rate settings I think reaching the latency requirement (and not doing mixing) can certainly be done in user-space, so I still read the "need to access harwdare directly" more as "existing user-space components are not suitable for my needs" than a strict requirement for doing this in kernel. But alas, I get your context and I understand these constraints are real for you as well. Just wanted to make sure you have taken good enough look at user-space options. > All that's left for me to do is to connect pipe the data directly to the > driver for the my sound card. Question is can it be done? And if so, > where do I even start? How can I import snd_usb_audio and open it so it > appears a the typical snd_card struct? The longrunning trend has been to push stuff like this to user-space, so you will not find much examples. I haven't done this myself, so I'll let other people make recommendations. I'd start by looking at ALSA OSS emulation and the virtualized drivers we have (like sound/xen/). Br, Kai

1 0

Re: [alsa-devel] kernel freeze regression relating to imx-sdma
by Kevin Groeneveld 04 Oct '19

04 Oct '19

Hi Lucas, Thanks for the feedback. >-----Original Message----- >From: Lucas Stach [mailto:l.stach@pengutronix.de] >Sent: October 3, 2019 5:26 PM >To: Kevin Groeneveld; dmaengine(a)vger.kernel.org >Subject: Re: kernel freeze regression relating to imx-sdma > >Hi Kevin, > >Am Donnerstag, den 03.10.2019, 21:11 +0000 schrieb Kevin Groeneveld: >> We have recently come across an issue which can cause the linux kernel to >> freeze on an iMX6 based system. After bisecting the issue I have found the >> issue is related to the following imx-sdma commits: >> >> 64068853bc77786d1a28abb4087d6a3e93aedbe2 dmaengine: imx-sdma: use GFP_NOWAIT for dma descriptor allocations >> b8603d2a5795c42f78998e70dc792336e0dc20c9 dmaengine: imx-sdma: implement channel termination via worker >> ebb853b1bd5f659b92c71dc6a9de44cfc37c78c0 Revert "dmaengine: imx-sdma: alloclate bd memory from dma pool" >> c06abca69218ac42fa58d1ba7a7b0d9bab5f1b18 Revert "dmaengine: imx-sdma: Use GFP_NOWAIT for dma allocations" >> >> If I revert these four commits the freeze does not happen. The issue is still >> present up to at least v5.3. >> >> The problem seems to be related to overruns when doing an SPDIF capture, >> particularly when there is not actually a valid SPDIF input to the iMX6. When >> the issue occurs the following is (usually) printed on the console (using a v5.0 >> based kernel): >> >> [ 1251.944877] rcu: INFO: rcu_sched detected stalls on CPUs/tasks: >> [ 1251.950823] rcu: (detected by 0, t=2102 jiffies, g=220425, q=0) >> [ 1251.956866] rcu: All QSes seen, last rcu_sched kthread activity 2102 (95117-93015), jiffies_till_next_fqs=1, root ->qsmask 0x0 >> [ 1251.968266] arecord S 0 1916 1908 0x00000002 >> [ 1251.973816] [<8010dcec>] (unwind_backtrace) from [<8010a4c4>] (show_stack+0x10/0x14) >> [ 1251.981596] [<8010a4c4>] (show_stack) from [<8016b314>] (rcu_check_callbacks+0x57c/0x5d8) >> [ 1251.989806] [<8016b314>] (rcu_check_callbacks) from [<8016fe84>] (update_process_times+0x30/0x5c) >> [ 1251.998709] [<8016fe84>] (update_process_times) from [<8017e960>] (tick_sched_timer+0x44/0x94) >> [ 1252.007353] [<8017e960>] (tick_sched_timer) from [<80170a84>] (__hrtimer_run_queues+0xd8/0x160) >> [ 1252.016081] [<80170a84>] (__hrtimer_run_queues) from [<8017103c>] (hrtimer_interrupt+0xdc/0x284) >> [ 1252.024891] [<8017103c>] (hrtimer_interrupt) from [<8010d6c0>] (twd_handler+0x2c/0x38) >> [ 1252.032839] [<8010d6c0>] (twd_handler) from [<80160b34>] (handle_percpu_devid_irq+0x70/0x11c) >> [ 1252.041401] [<80160b34>] (handle_percpu_devid_irq) from [<8015c234>] (generic_handle_irq+0x18/0x28) >> [ 1252.050480] [<8015c234>] (generic_handle_irq) from [<8015c814>] (__handle_domain_irq+0xa0/0xb4) >> [ 1252.059211] [<8015c814>] (__handle_domain_irq) from [<8039098c>] (gic_handle_irq+0x58/0x90) >> [ 1252.067590] [<8039098c>] (gic_handle_irq) from [<80101a0c>] (__irq_svc+0x6c/0x90) >> [ 1252.075085] Exception stack(0x848afe30 to 0x848afe78) >> [ 1252.080155] fe20: 8decf200 00000000 00000000 80b84ac8 >> [ 1252.088354] fe40: 00000000 8decf200 892f7800 00001589 892f7800 00000000 00000000 848ae000 >> [ 1252.096549] fe60: 8d8cf000 848afe80 805721e8 8056aae4 60000013 ffffffff >> [ 1252.103200] [<80101a0c>] (__irq_svc) from [<8056aae4>] (__snd_pcm_stream_unlock_mode+0xc0/0xd8) >> [ 1252.111925] [<8056aae4>] (__snd_pcm_stream_unlock_mode) from [<00000000>] ( (null)) >> [ 1252.119695] rcu: rcu_sched kthread starved for 2102 jiffies! g220425 f0x2 RCU_GP_WAIT_FQS(5) ->state=0x0 ->cpu=0 >> [ 1252.129877] rcu: RCU grace-period kthread stack dump: >> [ 1252.134938] rcu_sched R running task 0 10 2 0x00000000 >> [ 1252.142034] [<806eee5c>] (__schedule) from [<806eef80>] (schedule+0x90/0xa0) >> [ 1252.149116] [<806eef80>] (schedule) from [<806f2244>] (schedule_timeout+0x1f0/0x238) >> [ 1252.156891] [<806f2244>] (schedule_timeout) from [<8016940c>] (rcu_gp_kthread+0x550/0x8f0) >> [ 1252.165183] [<8016940c>] (rcu_gp_kthread) from [<8013a358>] (kthread+0x110/0x128) >> [ 1252.172693] [<8013a358>] (kthread) from [<801010e8>] (ret_from_fork+0x14/0x2c) >> [ 1252.179925] Exception stack(0x8f877fb0 to 0x8f877ff8) >> [ 1252.184991] 7fa0: 00000000 00000000 00000000 00000000 >> [ 1252.193187] 7fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 >> [ 1252.201380] 7fe0: 00000000 00000000 00000000 00000000 00000013 00000000 >> >> I suspect the issue may not actually be a bug with imx-sdma but a >> race condition elsewhere that is exposed by the four mentioned >> commits. >> >> I did notice that when an SPDIF overrun occurs >> dmaengine_terminate_async is called but dmaengine_synchronize is >> not. Should these calls always be paired? Or is it okay to call >> dmaengine_terminate_async without dmaengine_synchronize in some >> cases? > >Yes, it seems this uncovered a bug in the SPDIF driver. Before those >commits the dmaengine_terminate_async in the SDMA driver wasn't really >async, now it is. The API requires that you call dmaengine_synchronize >eventually after the terminate_async before starting any new operations >on the dmaengine. It doesn't necessarily need to be in the same >codepath, but the the async terminate always needs to be paired up with >a synchronize. I don't think the bug is limited to SPDIF. The calls to dmaengine_terminate_async are in sound/core/pcm_dmaengine.c. When the ALSA device is closed it does call dmaengine_synchronize. But when there is an overrun it does not. It was commit bc0e7345168c0f7483d2d1da86285d89136417cd (by Lars-Peter Clausen) where the call to dmaengine_synchronize was added for closing an ALSA device. It seems the overrun case was overlooked? I think maybe I should copy the ALSA list on this as well. I have also found another manifestation of the imx-sdma change which may be useful for others trying to debug this. If I run the following command: arecord -Dhw:4,0 -fS24_LE -c2 -r44100 --buffer-size 100 > /dev/null With kernel v4.20-rc5 I get an endless stream of overrun messages. With kernel v4.20-rc6 (which includes the four imx-sdma commits mentioned above) I get a few overrun messages but then arecord exits with "arecord: pcm_read:2096: read error: Input/output error" Best regards, Kevin

1 0

[alsa-devel] Custom Kernel driver to open and use another ALSA PCM
by ed nwave 04 Oct '19

04 Oct '19

Hello, A little bit of an unorthodox request but I need my custom PCM driver to connect directly to another PCM driver if possible. I have an application that already sees my custom PCM driver as the connected sound card, but I also need to pipe the PCM data within my driver after altering it out to another PCM of my choice ( my actual sound card ). As an example, my PCM driver acts as a sound card connected to a microphone capture application, within the my custom PCM I've added various "dalek" and other voice changing alterations and now I need to output this stream into my actual sound card. The sound card I need my driver to import or connect to is a DAC over USB and I believe the loaded kernel module is snd_sub_audio which I can see via lsmod. It's PCM hw:1,0. So is this possible? An if so, how? Even the most basic information would be appreciated. Thanks.

2 1

[alsa-devel] Making alsamixer configurable, adding mouse support + minor fixes ...
by GitHub pull_request - opened 03 Oct '19

03 Oct '19

alsa-project/alsa-utils pull request #28 was opened from braph: ... continued from PR #26 ... Request URL : https://github.com/alsa-project/alsa-utils/pull/28 Patch URL : https://github.com/alsa-project/alsa-utils/pull/28.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH v2] FROMLIST: drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 03 Oct '19

03 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v1 to v2: - Use audio_lock to protect audio clock. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index aa7efd4da1c8..749d8e4c535b 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -1982,6 +1982,17 @@ static void hdmi_disable_overflow_interrupts(struct dw_hdmi *hdmi) HDMI_IH_MUTE_FC_STAT2); } +static void dw_hdmi_audio_restore(struct dw_hdmi *hdmi) +{ + unsigned long flags; + + spin_lock_irqsave(&hdmi->audio_lock, flags); + + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); + + spin_unlock_irqrestore(&hdmi->audio_lock, flags); +} + static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) { int ret; @@ -2045,7 +2056,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + dw_hdmi_audio_restore(hdmi); } /* not for DVI mode */ -- 2.23.0.444.g18eeb5a265-goog

1 0

Re: [alsa-devel] [External] Re: alsa-lib: Add ucm support for whiskeylake sof-skl_hda_card audio
by Pierre-Louis Bossart 03 Oct '19

03 Oct '19

On 9/19/19 9:54 AM, Mark Pearson wrote: >> >> Indeed UCM is required for all cases where SOF and PulseAudio are used. >> >> Our thinking was however to add this UCM file to the new repository outside >> of alsa-lib [1]. There is an on-going thread started by Jaroslav to move those >> files and relicense them as BSD-3-Clause [2] >> >> [1] >> https://mailman.alsa-project.org/pipermail/alsa-devel/2019- >> July/153257.html >> [2] >> https://mailman.alsa-project.org/pipermail/alsa-devel/2019- >> September/155246.html > > Thanks Pierre. > > Do we have any approximate timelines on when and how this will happen? (I'm new to this) > > One of my main aims is that we have a customer using Debian and one of our platforms that require this change - I need to make sure I understand how this would roll out and what actions they need to take in the meantime if it's not going to be available in Debian. I think the first order would be to have the file cleaned-up, with its Intel origin clearly stated with a signed-off-by tag. Then once this is done, the Debian package creation needs to be handled (using either the ALSA repo or the cloned version on SOF GitHub). I don't have any experience with Debian packages so can't really comment on the effort it would take.

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[alsa-devel] [PATCH v2 00/10] Basic sound support for Arndale board / wm8994 updates
by Sylwester Nawrocki 02 Oct '19

02 Oct '19

This patch series adds basic audio support for Exynos5250 SoC based Arndale board, the Bluetooth receiver source and HDMI output are not covered yet. There is also one fix for wm8994 driver related to WM1811 CODEC and wm8994 updates to handle MCLK clocks, similar to patches: ae1ea48c5c59 ("ASoC: arizona: Add gating for source clocks of the FLLs") 7a4413d0dc96 ("ASoC: arizona: Add gating for clock when used for direct MCLK") The main changes since v1: - Fixed bug in "ASoC: wm8994: Add support for MCLKn clock gating" patch (thanks to Charles), - dropped the MFD part patch and moved getting of clocks to the CODEC driver, - Added missing Kconfig entries to patch "ASoC: samsung: arndale: Add support for WM1811 CODEC", - Added a patch renaming the driver and a patch for exynos_defconfig. Sylwester Nawrocki (10): ASoC: wm8994: Do not register inapplicable controls for WM1811 ASoC: samsung: arndale: Add missing OF node dereferencing ASoC: wm8994: Add support for setting MCLK clock rate ASoC: wm8994: Add support for MCLKn clock gating ASoC: samsung: arndale: Simplify DAI link initialization ASoC: dt-bindings: Document "samsung,arndale-wm1811" compatible ASoC: samsung: arndale: Add support for WM1811 CODEC ASoC: samsung: Rename Arndale card driver ARM: dts: arndale: Add audio support (WM1811 CODEC boards) ARM: exynos_defconfig: Enable Arndale audio driver .../devicetree/bindings/sound/arndale.txt | 5 +- arch/arm/boot/dts/exynos5250-arndale.dts | 27 ++- arch/arm/configs/exynos_defconfig | 1 + sound/soc/codecs/wm8994.c | 199 +++++++++++++--- sound/soc/codecs/wm8994.h | 10 +- sound/soc/samsung/Kconfig | 12 +- sound/soc/samsung/Makefile | 4 +- sound/soc/samsung/arndale.c | 217 ++++++++++++++++++ sound/soc/samsung/arndale_rt5631.c | 138 ----------- 9 files changed, 438 insertions(+), 175 deletions(-) create mode 100644 sound/soc/samsung/arndale.c delete mode 100644 sound/soc/samsung/arndale_rt5631.c -- 2.17.1

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[alsa-devel] [PATCH] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 02 Oct '19

02 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems like proposed in the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 35 ++++++++++++ sound/hda/Kconfig | 10 +++- sound/hda/Makefile | 7 ++- sound/hda/intel-dsp-config.c | 97 ++++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +--- sound/pci/hda/hda_intel.c | 49 +++++----------- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 ++----- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 ++ 11 files changed, 173 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..700f86282a83 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,35 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_NODSP, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..c593253c9829 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,8 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +ifneq ($(CONFIG_ACPI),) + snd-intel-dspcfg-objs += intel-nhlt.o +endif +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..1abbf74ba643 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,97 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <sound/core.h> +#include <linux/pci.h> +#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=noDSP, 2=legacy, 3=SST, 4=SOF)"); + +static const u16 sof_skl_table[] = { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + 0x02c8, /* Cometlake-LP */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + 0x06c8, /* Cometlake-H */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + 0x3198, /* Geminilake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + 0x5a98, /* Broxton-P (Appololake) */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + 0x9dc8, /* Cannonlake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + 0xa348, /* Coffelake */ +#endif +}; + +static int snd_intel_dsp_check_device(u16 device, const u16 *table, u32 len) +{ + for (; len > 0; len--, table++) { + if (*table == device) + return 1; + } + return 0; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + if (snd_intel_dsp_check_device(pci->device, sof_skl_table, ARRAY_SIZE(sof_skl_table))) { + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_NODSP; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* DMIC check for Skylake+ */ + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + + return SND_INTEL_DSP_DRIVER_ANY; +} + +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..c76c2deea47c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool no_dsp_driver; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,8 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(no_dsp_driver, bool, 0444); +MODULE_PARM_DESC(no_dsp_driver, "Do not return from probe when another DSP driver claims device"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2020,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2037,17 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (!no_dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_NODSP && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2061,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

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[alsa-devel] [PATCH 1/2] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Maciej Falkowski 02 Oct '19

02 Oct '19

Convert Samsung I2S controller to newer dt-schema format Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ------------- .../bindings/sound/samsung-i2s.yaml | 119 ++++++++++++++++++ 2 files changed, 119 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..59dc76035cb4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sangbeom Kim <sbkim73(a)samsung.com> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with + secondary fifo, s/w reset control and internal mux for root clk src. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for + playback, stereo channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + + samsung,exynos7-i2s: with all the available features of exynos5 i2s. + + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - "samsung,s3c6410-i2s" + - "samsung,s5pv210-i2s" + - "samsung,exynos5420-i2s" + - "samsung,exynos7-i2s" + - "samsung,exynos7-i2s1" + + reg: + maxItems: 1 + + dmas: + description: list of DMA controller phandle and DMA request line ordered pairs. + + dma-names: + description: | + identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: + - const: iis + - const: i2s_opclk0 + - items: + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + description: | + "iis" is the i2s bus clock. + For i2s1 and i2s2 - "iis", "i2s_opclk0" + For i2s0 - "iis", "i2s_opclk0", "i2s_opclk1" + + "#clock-cells": + const: 1 + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + sub system(used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10 + &pdma0 9 + &pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss 0>, // EXYNOS_I2S_BUS + <&clock_audss 0>, // EXYNOS_I2S_BUS + <&clock_audss 0>; // EXYNOS_SCLK_I2S + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; + -- 2.17.1

5 19

[alsa-devel] Period size setting is ignored when dmix is used
by Andrea Narciso - ART S.p.A. 02 Oct '19

02 Oct '19

I would like to set different period size (or time) values for different ALSA applications, in order to optimize latency and processor load based on specific application needs. For example, a telephony application will use a shorter period size/time than a media player application, so that low latency does not affect the quality of the call. I found out that if dmix plugin is used in the ALSA path, period size/time settings are completely ignored and forced to default values (supposedly read from the ALSA driver of the HW card). For example, not using dmix, period size value is correctly accepted (but of course I cannot play audio from multiple applications at the same time): aplay -vv --period-size=4096 -D plug:hw <WAV file name> Plug PCM: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 ... period_size : 4096 ... Adding dmix instead, I can play audio from multiple applications at the same time, but period size is ignored and I cannot therefore set different latencies for different applications: aplay -vv --period-size=4096 -D plug:dmix <WAV file name> Plug PCM: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 ... period_size : 1024 ... Am I missing something? Is there a way to implement mixing + different latency settings in ALSA applications, without using an audio server? Thanks Andrea

2 1

[alsa-devel] [PATCH] ASoC: atmel: fix atmel_ssc_set_audio link failure
by Arnd Bergmann 02 Oct '19

02 Oct '19

The ssc audio driver can call into both pdc and dma backends. With the latest rework, the logic to do this in a safe way avoiding link errors was removed, bringing back link errors that were fixed long ago in commit 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") such as sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': atmel_ssc_dai.c:(.text+0xac): undefined reference to `atmel_pcm_pdc_platform_register' Fix it this time using Makefile hacks and a comment to prevent this from accidentally getting removed again rather than Kconfig hacks. Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/atmel/Kconfig | 4 ++-- sound/soc/atmel/Makefile | 10 ++++++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c229ed82..25c31bf64936 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -10,11 +10,11 @@ config SND_ATMEL_SOC if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC - tristate + bool depends on HAS_DMA config SND_ATMEL_SOC_DMA - tristate + bool select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 1f6890ed3738..c7d2989791be 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -6,8 +6,14 @@ snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o snd-soc-atmel-i2s-objs := atmel-i2s.o snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o -obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o -obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o +# pdc and dma need to both be built-in if any user of +# ssc is built-in. +ifdef CONFIG_SND_ATMEL_SOC_PDC +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-pdc.o +endif +ifdef CONFIG_SND_ATMEL_SOC_DMA +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-dma.o +endif obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o obj-$(CONFIG_SND_ATMEL_SOC_I2S) += snd-soc-atmel-i2s.o obj-$(CONFIG_SND_MCHP_SOC_I2S_MCC) += snd-soc-mchp-i2s-mcc.o -- 2.20.0

4 6

[alsa-devel] [PATCH v3 RESEND 1/3] ASoC: samsung: arndale: Simplify DAI link initialization
by Sylwester Nawrocki 02 Oct '19

02 Oct '19

There is only one DAI link so we can drop an unnecessary loop statement. Use card->dai_link in place of direct static arndale_rt5631_dai[] array dereference as a prerequisite for adding support for other CODECs. Unnecessary assignment of dai_link->codecs->name to NULL is removed. Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since v1: - rebased due to reordering patches in the series --- sound/soc/samsung/arndale_rt5631.c | 42 +++++++++++++----------------- 1 file changed, 18 insertions(+), 24 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index fd8c6642fb0d..004c84fafad9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -88,36 +88,30 @@ static void arndale_put_of_nodes(struct snd_soc_card *card) static int arndale_audio_probe(struct platform_device *pdev) { - int n, ret; struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &arndale_rt5631; + struct snd_soc_dai_link *dai_link; + int ret; card->dev = &pdev->dev; + dai_link = card->dai_link; + + dai_link->cpus->of_node = of_parse_phandle(np, "samsung,audio-cpu", 0); + if (!dai_link->cpus->of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-cpu' missing or invalid\n"); + return -EINVAL; + } - for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) { - if (!arndale_rt5631_dai[n].cpus->dai_name) { - arndale_rt5631_dai[n].cpus->of_node = of_parse_phandle(np, - "samsung,audio-cpu", n); - - if (!arndale_rt5631_dai[n].cpus->of_node) { - dev_err(&pdev->dev, - "Property 'samsung,audio-cpu' missing or invalid\n"); - return -EINVAL; - } - } - if (!arndale_rt5631_dai[n].platforms->name) - arndale_rt5631_dai[n].platforms->of_node = - arndale_rt5631_dai[n].cpus->of_node; - - arndale_rt5631_dai[n].codecs->name = NULL; - arndale_rt5631_dai[n].codecs->of_node = of_parse_phandle(np, - "samsung,audio-codec", n); - if (!arndale_rt5631_dai[0].codecs->of_node) { - dev_err(&pdev->dev, + if (!dai_link->platforms->name) + dai_link->platforms->of_node = dai_link->cpus->of_node; + + dai_link->codecs->of_node = of_parse_phandle(np, "samsung,audio-codec", 0); + if (!dai_link->codecs->of_node) { + dev_err(&pdev->dev, "Property 'samsung,audio-codec' missing or invalid\n"); - ret = -EINVAL; - goto err_put_of_nodes; - } + ret = -EINVAL; + goto err_put_of_nodes; } ret = devm_snd_soc_register_card(card->dev, card); -- 2.17.1

4 8

[alsa-devel] [PATCH v2 1/2] ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver
by Nuno Sá 02 Oct '19

02 Oct '19

This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be eight microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- Changes in v2: * Use C++ coment style on the entire comment in the .c files; * Do not use ternery operator; * Do not use `devm_regulator_get_optional` as the device always requires power; * Let DAPM suspend/resume the device. MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 28 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau7118-hw.c | 43 +++ sound/soc/codecs/adau7118-i2c.c | 82 +++++ sound/soc/codecs/adau7118.c | 586 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau7118.h | 24 ++ 7 files changed, 770 insertions(+) create mode 100644 sound/soc/codecs/adau7118-hw.c create mode 100644 sound/soc/codecs/adau7118-i2c.c create mode 100644 sound/soc/codecs/adau7118.c create mode 100644 sound/soc/codecs/adau7118.h diff --git a/MAINTAINERS b/MAINTAINERS index 296de2b51c83..8ba47cac8e83 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1002,6 +1002,7 @@ F: drivers/media/i2c/adv7842* ANALOG DEVICES INC ASOC CODEC DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> +M: Nuno Sá <nuno.sa(a)analog.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac95785493..ae213c6392ec 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADAU7002 + select SND_SOC_ADAU7118_I2C if I2C + select SND_SOC_ADAU7118_HW select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4118 if I2C @@ -396,6 +398,32 @@ config SND_SOC_ADAU1977_I2C config SND_SOC_ADAU7002 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter" +config SND_SOC_ADAU7118 + tristate + +config SND_SOC_ADAU7118_HW + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - HW Mode" + select SND_SOC_ADAU7118 + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter. In this mode, the device works in standalone mode which + means that there is no bus to comunicate with it. Stereo mode is not + supported in this mode. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-hw. + +config SND_SOC_ADAU7118_I2C + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + select SND_SOC_ADAU7118 + select REGMAP_I2C + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter over I2C. This gives full support over the device. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-i2c. + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 73b2d5982dcb..2118407a1dd5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,9 @@ snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adau7002-objs := adau7002.o +snd-soc-adau7118-objs := adau7118.o +snd-soc-adau7118-i2c-objs := adau7118-i2c.o +snd-soc-adau7118-hw-objs := adau7118-hw.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -305,6 +308,9 @@ obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o +obj-$(CONFIG_SND_SOC_ADAU7118) += snd-soc-adau7118.o +obj-$(CONFIG_SND_SOC_ADAU7118_I2C) += snd-soc-adau7118-i2c.o +obj-$(CONFIG_SND_SOC_ADAU7118_HW) += snd-soc-adau7118-hw.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau7118-hw.c b/sound/soc/codecs/adau7118-hw.c new file mode 100644 index 000000000000..45a5d2dcc0f2 --- /dev/null +++ b/sound/soc/codecs/adau7118-hw.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter Standalone Hw +// driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> + +#include "adau7118.h" + +static int adau7118_probe_hw(struct platform_device *pdev) +{ + return adau7118_probe(&pdev->dev, NULL, true); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct platform_device_id adau7118_id[] = { + { .name = "adau7118" }, + { } +}; +MODULE_DEVICE_TABLE(platform, adau7118_id); + +static struct platform_driver adau7118_driver_hw = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_hw, + .id_table = adau7118_id, +}; +module_platform_driver(adau7118_driver_hw); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver for standalone hw mode"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c new file mode 100644 index 000000000000..a8211362fe82 --- /dev/null +++ b/sound/soc/codecs/adau7118-i2c.c @@ -0,0 +1,82 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "adau7118.h" + +static const struct reg_default adau7118_reg_defaults[] = { + { ADAU7118_REG_VENDOR_ID, 0x41 }, + { ADAU7118_REG_DEVICE_ID1, 0x71 }, + { ADAU7118_REG_DEVICE_ID2, 0x18 }, + { ADAU7118_REG_REVISION_ID, 0x00 }, + { ADAU7118_REG_ENABLES, 0x3F }, + { ADAU7118_REG_DEC_RATIO_CLK_MAP, 0xC0 }, + { ADAU7118_REG_HPF_CONTROL, 0xD0 }, + { ADAU7118_REG_SPT_CTRL1, 0x41 }, + { ADAU7118_REG_SPT_CTRL2, 0x00 }, + { ADAU7118_REG_SPT_CX(0), 0x01 }, + { ADAU7118_REG_SPT_CX(1), 0x11 }, + { ADAU7118_REG_SPT_CX(2), 0x21 }, + { ADAU7118_REG_SPT_CX(3), 0x31 }, + { ADAU7118_REG_SPT_CX(4), 0x41 }, + { ADAU7118_REG_SPT_CX(5), 0x51 }, + { ADAU7118_REG_SPT_CX(6), 0x61 }, + { ADAU7118_REG_SPT_CX(7), 0x71 }, + { ADAU7118_REG_DRIVE_STRENGTH, 0x2a }, + { ADAU7118_REG_RESET, 0x00 }, +}; + +static const struct regmap_config adau7118_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adau7118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .max_register = ADAU7118_REG_RESET, +}; + +static int adau7118_probe_i2c(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *map; + + map = devm_regmap_init_i2c(i2c, &adau7118_regmap_config); + if (IS_ERR(map)) { + dev_err(&i2c->dev, "Failed to init regmap %ld\n", PTR_ERR(map)); + return PTR_ERR(map); + } + + return adau7118_probe(&i2c->dev, map, false); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct i2c_device_id adau7118_id[] = { + {"adau7118", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, adau7118_id); + +static struct i2c_driver adau7118_driver = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_i2c, + .id_table = adau7118_id, +}; +module_i2c_driver(adau7118_driver); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c new file mode 100644 index 000000000000..bf5a5d75f81a --- /dev/null +++ b/sound/soc/codecs/adau7118.c @@ -0,0 +1,586 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/bitfield.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau7118.h" + +#define ADAU7118_DEC_RATIO_MASK GENMASK(1, 0) +#define ADAU7118_DEC_RATIO(x) FIELD_PREP(ADAU7118_DEC_RATIO_MASK, x) +#define ADAU7118_CLK_MAP_MASK GENMASK(7, 4) +#define ADAU7118_SLOT_WIDTH_MASK GENMASK(5, 4) +#define ADAU7118_SLOT_WIDTH(x) FIELD_PREP(ADAU7118_SLOT_WIDTH_MASK, x) +#define ADAU7118_TRISTATE_MASK BIT(6) +#define ADAU7118_TRISTATE(x) FIELD_PREP(ADAU7118_TRISTATE_MASK, x) +#define ADAU7118_DATA_FMT_MASK GENMASK(3, 1) +#define ADAU7118_DATA_FMT(x) FIELD_PREP(ADAU7118_DATA_FMT_MASK, x) +#define ADAU7118_SAI_MODE_MASK BIT(0) +#define ADAU7118_SAI_MODE(x) FIELD_PREP(ADAU7118_SAI_MODE_MASK, x) +#define ADAU7118_LRCLK_BCLK_POL_MASK GENMASK(1, 0) +#define ADAU7118_LRCLK_BCLK_POL(x) \ + FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) +#define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) +#define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_FULL_SOFT_R_MASK BIT(1) +#define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) + +struct adau7118_data { + struct regmap *map; + struct device *dev; + struct regulator *iovdd; + struct regulator *dvdd; + u32 slot_width; + u32 slots; + bool hw_mode; + bool right_j; +}; + +/* Input Enable */ +static const struct snd_kcontrol_new adau7118_dapm_pdm_control[4] = { + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 0, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 1, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 2, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_sw[] = { + /* Input Enable Switches */ + SND_SOC_DAPM_SWITCH("PDM0", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[0]), + SND_SOC_DAPM_SWITCH("PDM1", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[1]), + SND_SOC_DAPM_SWITCH("PDM2", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[2]), + SND_SOC_DAPM_SWITCH("PDM3", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[3]), + + /* PDM Clocks */ + SND_SOC_DAPM_SUPPLY("PDM_CLK0", ADAU7118_REG_ENABLES, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM_CLK1", ADAU7118_REG_ENABLES, 5, 0, NULL, 0), + + /* Output channels */ + SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, ADAU7118_REG_SPT_CX(0), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 0, ADAU7118_REG_SPT_CX(1), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 0, ADAU7118_REG_SPT_CX(2), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 0, ADAU7118_REG_SPT_CX(3), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 0, ADAU7118_REG_SPT_CX(4), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 0, ADAU7118_REG_SPT_CX(5), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX7", "Capture", 0, ADAU7118_REG_SPT_CX(6), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX8", "Capture", 0, ADAU7118_REG_SPT_CX(7), + 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_sw[] = { + { "PDM0", "Capture Switch", "PDM_DAT0" }, + { "PDM1", "Capture Switch", "PDM_DAT1" }, + { "PDM2", "Capture Switch", "PDM_DAT2" }, + { "PDM3", "Capture Switch", "PDM_DAT3" }, + { "AIF1TX1", NULL, "PDM0" }, + { "AIF1TX2", NULL, "PDM0" }, + { "AIF1TX3", NULL, "PDM1" }, + { "AIF1TX4", NULL, "PDM1" }, + { "AIF1TX5", NULL, "PDM2" }, + { "AIF1TX6", NULL, "PDM2" }, + { "AIF1TX7", NULL, "PDM3" }, + { "AIF1TX8", NULL, "PDM3" }, + { "Capture", NULL, "PDM_CLK0" }, + { "Capture", NULL, "PDM_CLK1" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_hw[] = { + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_hw[] = { + { "AIF1TX", NULL, "PDM_DAT0" }, + { "AIF1TX", NULL, "PDM_DAT1" }, + { "AIF1TX", NULL, "PDM_DAT2" }, + { "AIF1TX", NULL, "PDM_DAT3" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets[] = { + SND_SOC_DAPM_INPUT("PDM_DAT0"), + SND_SOC_DAPM_INPUT("PDM_DAT1"), + SND_SOC_DAPM_INPUT("PDM_DAT2"), + SND_SOC_DAPM_INPUT("PDM_DAT3"), +}; + +static int adau7118_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int chan, ret; + + dev_dbg(st->dev, "Set channel map, %d", tx_num); + + for (chan = 0; chan < tx_num; chan++) { + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CX(chan), + ADAU7118_SPT_SLOT_MASK, + ADAU7118_SPT_SLOT(tx_slot[chan])); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set format, fmt:%d\n", fmt); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(0)); + break; + case SND_SOC_DAIFMT_LEFT_J: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(1)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + st->right_j = true; + break; + default: + dev_err(st->dev, "Invalid format %d", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (ret < 0) + return ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(0); + break; + case SND_SOC_DAIFMT_NB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(2); + break; + case SND_SOC_DAIFMT_IB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(1); + break; + case SND_SOC_DAIFMT_IB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(3); + break; + default: + dev_err(st->dev, "Invalid Inv mask %d", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL2, + ADAU7118_LRCLK_BCLK_POL_MASK, + regval); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret; + + dev_dbg(st->dev, "Set tristate, %d\n", tristate); + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_TRISTATE_MASK, + ADAU7118_TRISTATE(tristate)); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set tdm, slots:%d width:%d\n", slots, slot_width); + + switch (slot_width) { + case 32: + regval = ADAU7118_SLOT_WIDTH(0); + break; + case 24: + regval = ADAU7118_SLOT_WIDTH(2); + break; + case 16: + regval = ADAU7118_SLOT_WIDTH(1); + break; + default: + dev_err(st->dev, "Invalid slot width:%d\n", slot_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SLOT_WIDTH_MASK, regval); + if (ret < 0) + return ret; + + st->slot_width = slot_width; + st->slots = slots; + + return 0; +} + +static int adau7118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + u32 data_width = params_width(params), slots_width; + int ret; + u32 regval; + + if (!st->slots) { + /* set stereo mode */ + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SAI_MODE_MASK, + ADAU7118_SAI_MODE(0)); + if (ret < 0) + return ret; + + slots_width = 32; + } else { + slots_width = st->slot_width; + } + + if (data_width > slots_width) { + dev_err(st->dev, "Invalid data_width:%d, slots_width:%d", + data_width, slots_width); + return -EINVAL; + } + + if (st->right_j) { + switch (slots_width - data_width) { + case 8: + /* delay bclck by 8 */ + regval = ADAU7118_DATA_FMT(2); + break; + case 12: + /* delay bclck by 12 */ + regval = ADAU7118_DATA_FMT(3); + break; + case 16: + /* delay bclck by 16 */ + regval = ADAU7118_DATA_FMT(4); + break; + default: + dev_err(st->dev, + "Cannot set right_j setting, slot_w:%d, data_w:%d\n", + slots_width, data_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + regval); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(st->dev, "Set bias level %d\n", level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* power on */ + ret = regulator_enable(st->iovdd); + if (ret) + return ret; + + /* there's no timing constraints before enabling dvdd */ + ret = regulator_enable(st->dvdd); + if (ret) { + regulator_disable(st->iovdd); + return ret; + } + + if (st->hw_mode) + return 0; + + regcache_cache_only(st->map, false); + /* sync cache */ + ret = snd_soc_component_cache_sync(component); + } + break; + case SND_SOC_BIAS_OFF: + /* power off */ + ret = regulator_disable(st->dvdd); + if (ret) + return ret; + + ret = regulator_disable(st->iovdd); + if (ret) + return ret; + + if (st->hw_mode) + return 0; + + /* cache only */ + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + + break; + } + + return ret; +} + +static int adau7118_component_probe(struct snd_soc_component *component) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret = 0; + + if (st->hw_mode) { + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_hw, + ARRAY_SIZE(adau7118_widgets_hw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_hw, + ARRAY_SIZE(adau7118_routes_hw)); + } else { + snd_soc_component_init_regmap(component, st->map); + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_sw, + ARRAY_SIZE(adau7118_widgets_sw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_sw, + ARRAY_SIZE(adau7118_routes_sw)); + } + + return ret; +} + +static const struct snd_soc_dai_ops adau7118_ops = { + .hw_params = adau7118_hw_params, + .set_channel_map = adau7118_set_channel_map, + .set_fmt = adau7118_set_fmt, + .set_tdm_slot = adau7118_set_tdm_slot, + .set_tristate = adau7118_set_tristate, +}; + +static struct snd_soc_dai_driver adau7118_dai = { + .name = "adau7118-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S20_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 192000, + .sig_bits = 24, + }, +}; + +static const struct snd_soc_component_driver adau7118_component_driver = { + .probe = adau7118_component_probe, + .set_bias_level = adau7118_set_bias_level, + .dapm_widgets = adau7118_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void adau7118_regulator_disable(void *data) +{ + struct adau7118_data *st = data; + int ret; + /* + * If we fail to disable DVDD, don't bother in trying IOVDD. We + * actually don't want to be left in the situation where DVDD + * is enabled and IOVDD is disabled. + */ + ret = regulator_disable(st->dvdd); + if (ret) + return; + + regulator_disable(st->iovdd); +} + +static int adau7118_regulator_setup(struct adau7118_data *st) +{ + st->iovdd = devm_regulator_get(st->dev, "IOVDD"); + if (IS_ERR(st->iovdd)) { + dev_err(st->dev, "Could not get iovdd: %ld\n", + PTR_ERR(st->iovdd)); + return PTR_ERR(st->iovdd); + } + + st->dvdd = devm_regulator_get(st->dev, "DVDD"); + if (IS_ERR(st->dvdd)) { + dev_err(st->dev, "Could not get dvdd: %ld\n", + PTR_ERR(st->dvdd)); + return PTR_ERR(st->dvdd); + } + /* just assume the device is in reset */ + if (!st->hw_mode) { + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + } + + return devm_add_action_or_reset(st->dev, adau7118_regulator_disable, + st); +} + +static int adau7118_parset_dt(const struct adau7118_data *st) +{ + int ret; + u32 dec_ratio = 0; + /* 4 inputs */ + u32 clk_map[4], regval; + + if (st->hw_mode) + return 0; + + ret = device_property_read_u32(st->dev, "adi,decimation-ratio", + &dec_ratio); + if (!ret) { + switch (dec_ratio) { + case 64: + regval = ADAU7118_DEC_RATIO(0); + break; + case 32: + regval = ADAU7118_DEC_RATIO(1); + break; + case 16: + regval = ADAU7118_DEC_RATIO(2); + break; + default: + dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_DEC_RATIO_MASK, regval); + if (ret) + return ret; + } + + ret = device_property_read_u32_array(st->dev, "adi,pdm-clk-map", + clk_map, ARRAY_SIZE(clk_map)); + if (!ret) { + int pdm; + u32 _clk_map = 0; + + for (pdm = 0; pdm < ARRAY_SIZE(clk_map); pdm++) + _clk_map |= (clk_map[pdm] << (pdm + 4)); + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_CLK_MAP_MASK, _clk_map); + if (ret) + return ret; + } + + return 0; +} + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode) +{ + struct adau7118_data *st; + int ret; + + st = devm_kzalloc(dev, sizeof(*st), GFP_KERNEL); + if (!st) + return -ENOMEM; + + st->dev = dev; + st->hw_mode = hw_mode; + dev_set_drvdata(dev, st); + + if (!hw_mode) { + st->map = map; + adau7118_dai.ops = &adau7118_ops; + /* + * Perform a full soft reset. This will set all register's + * with their reset values. + */ + ret = regmap_update_bits(map, ADAU7118_REG_RESET, + ADAU7118_FULL_SOFT_R_MASK, + ADAU7118_FULL_SOFT_R(1)); + if (ret) + return ret; + } + + ret = adau7118_parset_dt(st); + if (ret) + return ret; + + ret = adau7118_regulator_setup(st); + if (ret) + return ret; + + return devm_snd_soc_register_component(dev, + &adau7118_component_driver, + &adau7118_dai, 1); +} +EXPORT_SYMBOL_GPL(adau7118_probe); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.h b/sound/soc/codecs/adau7118.h new file mode 100644 index 000000000000..c65679a4dff1 --- /dev/null +++ b/sound/soc/codecs/adau7118.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _LINUX_ADAU7118_H +#define _LINUX_ADAU7118_H + +struct regmap; +struct device; + +/* register map */ +#define ADAU7118_REG_VENDOR_ID 0x00 +#define ADAU7118_REG_DEVICE_ID1 0x01 +#define ADAU7118_REG_DEVICE_ID2 0x02 +#define ADAU7118_REG_REVISION_ID 0x03 +#define ADAU7118_REG_ENABLES 0x04 +#define ADAU7118_REG_DEC_RATIO_CLK_MAP 0x05 +#define ADAU7118_REG_HPF_CONTROL 0x06 +#define ADAU7118_REG_SPT_CTRL1 0x07 +#define ADAU7118_REG_SPT_CTRL2 0x08 +#define ADAU7118_REG_SPT_CX(num) (0x09 + (num)) +#define ADAU7118_REG_DRIVE_STRENGTH 0x11 +#define ADAU7118_REG_RESET 0x12 + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode); + +#endif -- 2.23.0

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[alsa-devel] [PATCH 1/2] ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver
by Nuno Sá 02 Oct '19

02 Oct '19

This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be eight microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 28 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau7118-hw.c | 43 +++ sound/soc/codecs/adau7118-i2c.c | 82 +++++ sound/soc/codecs/adau7118.c | 621 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau7118.h | 24 ++ 7 files changed, 805 insertions(+) create mode 100644 sound/soc/codecs/adau7118-hw.c create mode 100644 sound/soc/codecs/adau7118-i2c.c create mode 100644 sound/soc/codecs/adau7118.c create mode 100644 sound/soc/codecs/adau7118.h diff --git a/MAINTAINERS b/MAINTAINERS index a50e97a63bc8..77c920559d82 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -985,6 +985,7 @@ F: drivers/media/i2c/adv7842* ANALOG DEVICES INC ASOC CODEC DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> +M: Nuno Sá <nuno.sa(a)analog.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f89a5346299..006b0686776c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADAU7002 + select SND_SOC_ADAU7118_I2C if I2C + select SND_SOC_ADAU7118_HW select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4118 if I2C @@ -392,6 +394,32 @@ config SND_SOC_ADAU1977_I2C config SND_SOC_ADAU7002 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter" +config SND_SOC_ADAU7118 + tristate + +config SND_SOC_ADAU7118_HW + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - HW Mode" + select SND_SOC_ADAU7118 + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter. In this mode, the device works in standalone mode which + means that there is no bus to comunicate with it. Stereo mode is not + supported in this mode. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-hw. + +config SND_SOC_ADAU7118_I2C + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + select SND_SOC_ADAU7118 + select REGMAP_I2C + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter over I2C. This gives full support over the device. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-i2c. + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b4bb8cf4325..974dc1eda797 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,9 @@ snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adau7002-objs := adau7002.o +snd-soc-adau7118-objs := adau7118.o +snd-soc-adau7118-i2c-objs := adau7118-i2c.o +snd-soc-adau7118-hw-objs := adau7118-hw.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -301,6 +304,9 @@ obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o +obj-$(CONFIG_SND_SOC_ADAU7118) += snd-soc-adau7118.o +obj-$(CONFIG_SND_SOC_ADAU7118_I2C) += snd-soc-adau7118-i2c.o +obj-$(CONFIG_SND_SOC_ADAU7118_HW) += snd-soc-adau7118-hw.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau7118-hw.c b/sound/soc/codecs/adau7118-hw.c new file mode 100644 index 000000000000..499dd48ea01b --- /dev/null +++ b/sound/soc/codecs/adau7118-hw.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter Standalone Hw + * driver + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> + +#include "adau7118.h" + +static int adau7118_probe_hw(struct platform_device *pdev) +{ + return adau7118_probe(&pdev->dev, NULL, true); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct platform_device_id adau7118_id[] = { + { .name = "adau7118" }, + { } +}; +MODULE_DEVICE_TABLE(platform, adau7118_id); + +static struct platform_driver adau7118_driver_hw = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_hw, + .id_table = adau7118_id, +}; +module_platform_driver(adau7118_driver_hw); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver for standalone hw mode"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c new file mode 100644 index 000000000000..1fd0b0287699 --- /dev/null +++ b/sound/soc/codecs/adau7118-i2c.c @@ -0,0 +1,82 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "adau7118.h" + +static const struct reg_default adau7118_reg_defaults[] = { + { ADAU7118_REG_VENDOR_ID, 0x41 }, + { ADAU7118_REG_DEVICE_ID1, 0x71 }, + { ADAU7118_REG_DEVICE_ID2, 0x18 }, + { ADAU7118_REG_REVISION_ID, 0x00 }, + { ADAU7118_REG_ENABLES, 0x3F }, + { ADAU7118_REG_DEC_RATIO_CLK_MAP, 0xC0 }, + { ADAU7118_REG_HPF_CONTROL, 0xD0 }, + { ADAU7118_REG_SPT_CTRL1, 0x41 }, + { ADAU7118_REG_SPT_CTRL2, 0x00 }, + { ADAU7118_REG_SPT_CX(0), 0x01 }, + { ADAU7118_REG_SPT_CX(1), 0x11 }, + { ADAU7118_REG_SPT_CX(2), 0x21 }, + { ADAU7118_REG_SPT_CX(3), 0x31 }, + { ADAU7118_REG_SPT_CX(4), 0x41 }, + { ADAU7118_REG_SPT_CX(5), 0x51 }, + { ADAU7118_REG_SPT_CX(6), 0x61 }, + { ADAU7118_REG_SPT_CX(7), 0x71 }, + { ADAU7118_REG_DRIVE_STRENGTH, 0x2a }, + { ADAU7118_REG_RESET, 0x00 }, +}; + +static const struct regmap_config adau7118_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adau7118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .max_register = ADAU7118_REG_RESET, +}; + +static int adau7118_probe_i2c(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *map; + + map = devm_regmap_init_i2c(i2c, &adau7118_regmap_config); + if (IS_ERR(map)) { + dev_err(&i2c->dev, "Failed to init regmap %ld\n", PTR_ERR(map)); + return PTR_ERR(map); + } + + return adau7118_probe(&i2c->dev, map, false); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct i2c_device_id adau7118_id[] = { + {"adau7118", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, adau7118_id); + +static struct i2c_driver adau7118_driver = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_i2c, + .id_table = adau7118_id, +}; +module_i2c_driver(adau7118_driver); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c new file mode 100644 index 000000000000..d6b90e31042f --- /dev/null +++ b/sound/soc/codecs/adau7118.c @@ -0,0 +1,621 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/bitfield.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau7118.h" + +#define ADAU7118_DEC_RATIO_MASK GENMASK(1, 0) +#define ADAU7118_DEC_RATIO(x) FIELD_PREP(ADAU7118_DEC_RATIO_MASK, x) +#define ADAU7118_CLK_MAP_MASK GENMASK(7, 4) +#define ADAU7118_SLOT_WIDTH_MASK GENMASK(5, 4) +#define ADAU7118_SLOT_WIDTH(x) FIELD_PREP(ADAU7118_SLOT_WIDTH_MASK, x) +#define ADAU7118_TRISTATE_MASK BIT(6) +#define ADAU7118_TRISTATE(x) FIELD_PREP(ADAU7118_TRISTATE_MASK, x) +#define ADAU7118_DATA_FMT_MASK GENMASK(3, 1) +#define ADAU7118_DATA_FMT(x) FIELD_PREP(ADAU7118_DATA_FMT_MASK, x) +#define ADAU7118_SAI_MODE_MASK BIT(0) +#define ADAU7118_SAI_MODE(x) FIELD_PREP(ADAU7118_SAI_MODE_MASK, x) +#define ADAU7118_LRCLK_BCLK_POL_MASK GENMASK(1, 0) +#define ADAU7118_LRCLK_BCLK_POL(x) \ + FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) +#define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) +#define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_FULL_SOFT_R_MASK BIT(1) +#define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) + +struct adau7118_data { + struct regmap *map; + struct device *dev; + struct regulator *iovdd; + struct regulator *dvdd; + u32 slot_width; + u32 slots; + bool hw_mode; + bool right_j; +}; + +/* Input Enable */ +static const struct snd_kcontrol_new adau7118_dapm_pdm_control[4] = { + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 0, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 1, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 2, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_sw[] = { + /* Input Enable Switches */ + SND_SOC_DAPM_SWITCH("PDM0", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[0]), + SND_SOC_DAPM_SWITCH("PDM1", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[1]), + SND_SOC_DAPM_SWITCH("PDM2", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[2]), + SND_SOC_DAPM_SWITCH("PDM3", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[3]), + + /* PDM Clocks */ + SND_SOC_DAPM_SUPPLY("PDM_CLK0", ADAU7118_REG_ENABLES, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM_CLK1", ADAU7118_REG_ENABLES, 5, 0, NULL, 0), + + /* Output channels */ + SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, ADAU7118_REG_SPT_CX(0), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 0, ADAU7118_REG_SPT_CX(1), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 0, ADAU7118_REG_SPT_CX(2), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 0, ADAU7118_REG_SPT_CX(3), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 0, ADAU7118_REG_SPT_CX(4), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 0, ADAU7118_REG_SPT_CX(5), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX7", "Capture", 0, ADAU7118_REG_SPT_CX(6), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX8", "Capture", 0, ADAU7118_REG_SPT_CX(7), + 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_sw[] = { + { "PDM0", "Capture Switch", "PDM_DAT0" }, + { "PDM1", "Capture Switch", "PDM_DAT1" }, + { "PDM2", "Capture Switch", "PDM_DAT2" }, + { "PDM3", "Capture Switch", "PDM_DAT3" }, + { "AIF1TX1", NULL, "PDM0" }, + { "AIF1TX2", NULL, "PDM0" }, + { "AIF1TX3", NULL, "PDM1" }, + { "AIF1TX4", NULL, "PDM1" }, + { "AIF1TX5", NULL, "PDM2" }, + { "AIF1TX6", NULL, "PDM2" }, + { "AIF1TX7", NULL, "PDM3" }, + { "AIF1TX8", NULL, "PDM3" }, + { "Capture", NULL, "PDM_CLK0" }, + { "Capture", NULL, "PDM_CLK1" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_hw[] = { + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_hw[] = { + { "AIF1TX", NULL, "PDM_DAT0" }, + { "AIF1TX", NULL, "PDM_DAT1" }, + { "AIF1TX", NULL, "PDM_DAT2" }, + { "AIF1TX", NULL, "PDM_DAT3" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets[] = { + SND_SOC_DAPM_INPUT("PDM_DAT0"), + SND_SOC_DAPM_INPUT("PDM_DAT1"), + SND_SOC_DAPM_INPUT("PDM_DAT2"), + SND_SOC_DAPM_INPUT("PDM_DAT3"), +}; + +static int adau7118_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int chan, ret; + + dev_dbg(st->dev, "Set channel map, %d", tx_num); + + for (chan = 0; chan < tx_num; chan++) { + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CX(chan), + ADAU7118_SPT_SLOT_MASK, + ADAU7118_SPT_SLOT(tx_slot[chan])); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set format, fmt:%d\n", fmt); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(0)); + break; + case SND_SOC_DAIFMT_LEFT_J: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(1)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + st->right_j = true; + break; + default: + dev_err(st->dev, "Invalid format %d", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (ret < 0) + return ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(0); + break; + case SND_SOC_DAIFMT_NB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(2); + break; + case SND_SOC_DAIFMT_IB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(1); + break; + case SND_SOC_DAIFMT_IB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(3); + break; + default: + dev_err(st->dev, "Invalid Inv mask %d", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL2, + ADAU7118_LRCLK_BCLK_POL_MASK, + regval); + + return ret < 0 ? ret : 0; +} + +static int adau7118_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret; + + dev_dbg(st->dev, "Set tristate, %d\n", tristate); + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_TRISTATE_MASK, + ADAU7118_TRISTATE(tristate)); + + return ret < 0 ? ret : 0; +} + +static int adau7118_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set tdm, slots:%d width:%d\n", slots, slot_width); + + switch (slot_width) { + case 32: + regval = ADAU7118_SLOT_WIDTH(0); + break; + case 24: + regval = ADAU7118_SLOT_WIDTH(2); + break; + case 16: + regval = ADAU7118_SLOT_WIDTH(1); + break; + default: + dev_err(st->dev, "Invalid slot width:%d\n", slot_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SLOT_WIDTH_MASK, regval); + if (ret < 0) + return ret; + + st->slot_width = slot_width; + st->slots = slots; + + return 0; +} + +static int adau7118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + u32 data_width = params_width(params), slots_width; + int ret; + u32 regval; + + if (!st->slots) { + /* set stereo mode */ + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SAI_MODE_MASK, + ADAU7118_SAI_MODE(0)); + if (ret < 0) + return ret; + + slots_width = 32; + } else { + slots_width = st->slot_width; + } + + if (data_width > slots_width) { + dev_err(st->dev, "Invalid data_width:%d, slots_width:%d", + data_width, slots_width); + return -EINVAL; + } + + if (st->right_j) { + switch (slots_width - data_width) { + case 8: + /* delay bclck by 8 */ + regval = ADAU7118_DATA_FMT(2); + break; + case 12: + /* delay bclck by 12 */ + regval = ADAU7118_DATA_FMT(3); + break; + case 16: + /* delay bclck by 16 */ + regval = ADAU7118_DATA_FMT(4); + break; + default: + dev_err(st->dev, + "Cannot set right_j setting, slot_w:%d, data_w:%d\n", + slots_width, data_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + regval); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(st->dev, "Set bias level %d\n", level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + if (!st->iovdd) + return 0; + + /* power on */ + ret = regulator_enable(st->iovdd); + if (ret) + return ret; + + /* there's no timing constraints before enabling dvdd */ + ret = regulator_enable(st->dvdd); + if (ret) { + regulator_disable(st->iovdd); + return ret; + } + + if (st->hw_mode) + return 0; + + regcache_cache_only(st->map, false); + /* sync cache */ + ret = snd_soc_component_cache_sync(component); + } + break; + case SND_SOC_BIAS_OFF: + /* power off */ + if (!st->iovdd) + return 0; + + ret = regulator_disable(st->dvdd); + if (ret) + return ret; + + ret = regulator_disable(st->iovdd); + if (ret) + return ret; + + if (st->hw_mode) + return 0; + + /* cache only */ + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + + break; + } + + return ret; +} + +static int adau7118_component_probe(struct snd_soc_component *component) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret = 0; + + if (st->hw_mode) { + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_hw, + ARRAY_SIZE(adau7118_widgets_hw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_hw, + ARRAY_SIZE(adau7118_routes_hw)); + } else { + snd_soc_component_init_regmap(component, st->map); + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_sw, + ARRAY_SIZE(adau7118_widgets_sw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_sw, + ARRAY_SIZE(adau7118_routes_sw)); + } + + return ret; +} + +#ifdef CONFIG_PM +static int adau7118_suspend(struct snd_soc_component *component) +{ + return snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); +} + +static int adau7118_resume(struct snd_soc_component *component) +{ + return snd_soc_component_force_bias_level(component, + SND_SOC_BIAS_STANDBY); +} +#else +#define adau7118_suspend NULL +#define adau7118_resume NULL +#endif + +static const struct snd_soc_dai_ops adau7118_ops = { + .hw_params = adau7118_hw_params, + .set_channel_map = adau7118_set_channel_map, + .set_fmt = adau7118_set_fmt, + .set_tdm_slot = adau7118_set_tdm_slot, + .set_tristate = adau7118_set_tristate, +}; + +static struct snd_soc_dai_driver adau7118_dai = { + .name = "adau7118-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S20_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 192000, + .sig_bits = 24, + }, +}; + +static const struct snd_soc_component_driver adau7118_component_driver = { + .probe = adau7118_component_probe, + .set_bias_level = adau7118_set_bias_level, + .suspend = adau7118_suspend, + .resume = adau7118_resume, + .dapm_widgets = adau7118_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void adau7118_regulator_disable(void *data) +{ + struct adau7118_data *st = data; + int ret; + /* + * If we fail to disable DVDD, don't bother in trying IOVDD. We + * actually don't want to be left in the situation where DVDD + * is enabled and IOVDD is disabled. + */ + ret = regulator_disable(st->dvdd); + if (ret) + return; + + regulator_disable(st->iovdd); +} + +static int adau7118_regulator_setup(struct adau7118_data *st) +{ + int ret = 0; + + st->iovdd = devm_regulator_get_optional(st->dev, "IOVDD"); + if (!IS_ERR(st->iovdd)) { + /* + * Only get dvdd if iovdd is present. Furthermore, + * if iovdd is present, dvdd must also be given. The two + * supplies should be controlled together since dvdd should + * not be applied before iovdd. + */ + st->dvdd = devm_regulator_get(st->dev, "DVDD"); + if (IS_ERR(st->dvdd)) { + dev_err(st->dev, "Could not get dvdd: %ld\n", + PTR_ERR(st->dvdd)); + return PTR_ERR(st->dvdd); + } + /* just assume the device is in reset */ + if (!st->hw_mode) { + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + } + + ret = devm_add_action_or_reset(st->dev, + adau7118_regulator_disable, st); + } else { + if (PTR_ERR(st->iovdd) == -ENODEV) { + st->iovdd = NULL; + } else { + dev_err(st->dev, "Failed to get IOVDD %ld\n", + PTR_ERR(st->iovdd)); + return PTR_ERR(st->iovdd); + } + } + + return ret; +} + +static int adau7118_parset_dt(const struct adau7118_data *st) +{ + int ret; + u32 dec_ratio = 0; + /* 4 inputs */ + u32 clk_map[4], regval; + + if (st->hw_mode) + return 0; + + ret = device_property_read_u32(st->dev, "adi,decimation-ratio", + &dec_ratio); + if (!ret) { + switch (dec_ratio) { + case 64: + regval = ADAU7118_DEC_RATIO(0); + break; + case 32: + regval = ADAU7118_DEC_RATIO(1); + break; + case 16: + regval = ADAU7118_DEC_RATIO(2); + break; + default: + dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_DEC_RATIO_MASK, regval); + if (ret) + return ret; + } + + ret = device_property_read_u32_array(st->dev, "adi,pdm-clk-map", + clk_map, ARRAY_SIZE(clk_map)); + if (!ret) { + int pdm; + u32 _clk_map = 0; + + for (pdm = 0; pdm < ARRAY_SIZE(clk_map); pdm++) + _clk_map |= (clk_map[pdm] << (pdm + 4)); + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_CLK_MAP_MASK, _clk_map); + if (ret) + return ret; + } + + return 0; +} + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode) +{ + struct adau7118_data *st; + int ret; + + st = devm_kzalloc(dev, sizeof(*st), GFP_KERNEL); + if (!st) + return -ENOMEM; + + st->dev = dev; + st->hw_mode = hw_mode; + dev_set_drvdata(dev, st); + + if (!hw_mode) { + st->map = map; + adau7118_dai.ops = &adau7118_ops; + /* + * Perform a full soft reset. This will set all register's + * with their reset values. + */ + ret = regmap_update_bits(map, ADAU7118_REG_RESET, + ADAU7118_FULL_SOFT_R_MASK, + ADAU7118_FULL_SOFT_R(1)); + if (ret) + return ret; + } + + ret = adau7118_parset_dt(st); + if (ret) + return ret; + + ret = adau7118_regulator_setup(st); + if (ret) + return ret; + + return devm_snd_soc_register_component(dev, + &adau7118_component_driver, + &adau7118_dai, 1); +} +EXPORT_SYMBOL_GPL(adau7118_probe); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.h b/sound/soc/codecs/adau7118.h new file mode 100644 index 000000000000..c65679a4dff1 --- /dev/null +++ b/sound/soc/codecs/adau7118.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _LINUX_ADAU7118_H +#define _LINUX_ADAU7118_H + +struct regmap; +struct device; + +/* register map */ +#define ADAU7118_REG_VENDOR_ID 0x00 +#define ADAU7118_REG_DEVICE_ID1 0x01 +#define ADAU7118_REG_DEVICE_ID2 0x02 +#define ADAU7118_REG_REVISION_ID 0x03 +#define ADAU7118_REG_ENABLES 0x04 +#define ADAU7118_REG_DEC_RATIO_CLK_MAP 0x05 +#define ADAU7118_REG_HPF_CONTROL 0x06 +#define ADAU7118_REG_SPT_CTRL1 0x07 +#define ADAU7118_REG_SPT_CTRL2 0x08 +#define ADAU7118_REG_SPT_CX(num) (0x09 + (num)) +#define ADAU7118_REG_DRIVE_STRENGTH 0x11 +#define ADAU7118_REG_RESET 0x12 + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode); + +#endif -- 2.23.0

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[alsa-devel] alsa acting up
by liebrecht＠grossmann-venter.com 02 Oct '19

02 Oct '19

See images in links. 1) "http://grossmann-venter.com/visitors/ug/alsa/alsa-pulse-181vsl_unplug.png" 2) "http://grossmann-venter.com/visitors/ug/alsa/alsa-dont recognize1818VSLpng" 1) As soon as pulseaudio is started Alsa says 1818VSL is unplugged....? 2) Clearly Presonus 1818VSL is found as lsusb output says, but alsa has no clue about it. How do I get alsa to recognise 181VSL as it is found by lsusb so it is presnet on the system. It used to work fine but alsa just stopped working with 1818VSL recently for no reason. I know this is a developers group, but please at least answer where I can get help. Alsa help is absolutely non-existing. I also posted a bug with no response in another thread. Thanks.

2 3

[alsa-devel] [PATCH 00/44] ASoC: don't use snd_pcm_ops
by Kuninori Morimoto 02 Oct '19

02 Oct '19

Hi Mark Current snd_soc_component_driver is using snd_pcm_ops. But, it is mainly for ALSA, thus, it not good much to ALSA SoC. For example .open case (1)(2), each callback can't know it is called for which component. Thus, it need to use using snd_soc_rtdcom_lookup() (3). --- ALSA SoC --- ... if (component->driver->ops && component->driver->ops->open) (1) return component->driver->ops->open(substream); ... --- driver --- (2) static int xxx_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; (3) struct snd_soc_component *component = snd_soc_rtdcom_lookup(..); ... } It works today, but, will not work in the future if we support multi CPU/Codec/Platform, because 1 rtd might have multiple components which have same driver name. In other words snd_soc_rtdcom_lookup() doesn't work correctly in the future. To solve this issue, each callback needs to be called with component. This patch copies each snd_pcm_ops member under component driver, and having "component" as parameter. --- ALSA SoC --- ... if (component->driver->open) => return component->driver->open(component, substream); ... --- driver --- => static int xxx_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { ... } Kuninori Morimoto (44): ASoC: soc-core: merge snd_pcm_ops member to component driver ASoC: soc-core: add new pcm_construct/pcm_destruct ASoC: soc-core: add snd_soc_pcm_lib_ioctl() ASoC: intel: sst-haswell-pcm: remove snd_pcm_ops ASoC: intel: skl-pcm: remove snd_pcm_ops ASoC: intel: sst-mfld-platform-pcm: remove snd_pcm_ops ASoC: intel: sst-baytrail-pcm: remove snd_pcm_ops ASoC: amd: acp3x-pcm-dma: remove snd_pcm_ops ASoC: amd: acp-pcm-dma: remove snd_pcm_ops ASoC: atmel: atmel-pcm-pdc: remove snd_pcm_ops ASoC: au1x: dbdma2: remove snd_pcm_ops ASoC: au1x: dma: remove snd_pcm_ops ASoC: xtensa: xtfpga-i2s: remove snd_pcm_ops ASoC: xilinx: xlnx_formatter_pcm: remove snd_pcm_ops ASoC: uniphier: aio-dma: remove snd_pcm_ops ASoC: txx9: txx9aclc: remove snd_pcm_ops ASoC: stm: stm32_adfsdm: remove snd_pcm_ops ASoC: sprd: sprd-pcm-dma: remove snd_pcm_ops ASoC: sof: pcm: remove snd_pcm_ops ASoC: sh: rsnd: remove snd_pcm_ops ASoC: sh: fsi: remove snd_pcm_ops ASoC: sh: dma-sh7760: remove snd_pcm_ops ASoC: sh: siu_pcm: remove snd_pcm_ops ASoC: samsung: idma: remove snd_pcm_ops ASoC: qcom: lpass-platform: remove snd_pcm_ops ASoC: qcom: q6routing: remove snd_pcm_ops ASoC: qcom: q6asm-dai: remove snd_pcm_ops ASoC: pxa: mmp-pcm: remove snd_pcm_ops ASoC: pxa: remove snd_pcm_ops ASoC: meson: remove snd_pcm_ops ASoC: mediatek: mtk-btcvsd: remove snd_pcm_ops ASoC: mediatek: remove snd_pcm_ops ASoC: kirkwood: kirkwood-dma: remove snd_pcm_ops ASoC: dwc: dwc-pcm: remove snd_pcm_ops ASoC: rt5514-spi: remove snd_pcm_ops ASoC: bcm: cygnus-pcm: remove snd_pcm_ops ASoC: fsl: imx-pcm-fiq: remove snd_pcm_ops ASoC: fsl: fsl_dma: remove snd_pcm_ops ASoC: fsl: mpc5200_dma: remove snd_pcm_ops ASoC: fsl: fsl_asrc_dma: remove snd_pcm_ops ASoC: fsl: fsl_dma: don't use snd_soc_rtdcom_lookup() ASoC: soc-generic-dmaengine-pcm: remove snd_pcm_ops ASoC: soc-utils: remove snd_pcm_ops ASoC: soc-component: remove snd_pcm_ops from component driver include/sound/pxa2xx-lib.h | 26 +++++- include/sound/soc-component.h | 41 ++++++++- include/sound/soc.h | 5 + sound/arm/pxa2xx-pcm-lib.c | 80 +++++++++++++--- sound/soc/amd/acp-pcm-dma.c | 63 ++++++------- sound/soc/amd/raven/acp3x-pcm-dma.c | 54 +++++------ sound/soc/atmel/atmel-pcm-pdc.c | 48 +++++----- sound/soc/au1x/dbdma2.c | 64 +++++++------ sound/soc/au1x/dma.c | 63 +++++++------ sound/soc/bcm/cygnus-pcm.c | 56 +++++------ sound/soc/codecs/rt5514-spi.c | 41 +++++---- sound/soc/dwc/dwc-pcm.c | 48 +++++----- sound/soc/fsl/fsl_asrc_dma.c | 56 +++++------ sound/soc/fsl/fsl_dma.c | 53 +++++------ sound/soc/fsl/imx-pcm-fiq.c | 56 ++++++----- sound/soc/fsl/mpc5200_dma.c | 51 ++++++----- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 30 +++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 52 +++++------ sound/soc/intel/haswell/sst-haswell-pcm.c | 62 +++++++------ sound/soc/intel/skylake/skl-pcm.c | 67 +++++++++----- sound/soc/kirkwood/kirkwood-dma.c | 50 +++++----- .../soc/mediatek/common/mtk-afe-platform-driver.c | 28 +++--- .../soc/mediatek/common/mtk-afe-platform-driver.h | 10 +- sound/soc/mediatek/common/mtk-btcvsd.c | 76 ++++++--------- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 11 ++- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 11 ++- sound/soc/meson/axg-fifo.c | 56 +++++------ sound/soc/meson/axg-fifo.h | 20 +++- sound/soc/meson/axg-frddr.c | 24 ++++- sound/soc/meson/axg-toddr.c | 24 ++++- sound/soc/pxa/mmp-pcm.c | 62 ++++++++----- sound/soc/pxa/pxa-ssp.c | 14 ++- sound/soc/pxa/pxa2xx-ac97.c | 14 ++- sound/soc/pxa/pxa2xx-i2s.c | 14 ++- sound/soc/pxa/pxa2xx-pcm.c | 14 ++- sound/soc/qcom/lpass-platform.c | 70 +++++++------- sound/soc/qcom/qdsp6/q6asm-dai.c | 71 +++++++------- sound/soc/qcom/qdsp6/q6routing.c | 14 +-- sound/soc/samsung/idma.c | 58 ++++++------ sound/soc/sh/dma-sh7760.c | 46 +++++----- sound/soc/sh/fsi.c | 31 ++++--- sound/soc/sh/rcar/core.c | 23 +++-- sound/soc/sh/siu_pcm.c | 44 +++++---- sound/soc/soc-component.c | 76 +++++++-------- sound/soc/soc-core.c | 12 +++ sound/soc/soc-generic-dmaengine-pcm.c | 102 ++++++++++----------- sound/soc/soc-pcm.c | 11 +-- sound/soc/soc-utils.c | 11 +-- sound/soc/sof/pcm.c | 88 +++++++++--------- sound/soc/sprd/sprd-pcm-dma.c | 74 +++++++-------- sound/soc/stm/stm32_adfsdm.c | 42 +++++---- sound/soc/txx9/txx9aclc.c | 48 +++++----- sound/soc/uniphier/aio-dma.c | 51 ++++++----- sound/soc/xilinx/xlnx_formatter_pcm.c | 54 +++++------ sound/soc/xtensa/xtfpga-i2s.c | 30 +++--- 55 files changed, 1311 insertions(+), 1089 deletions(-) -- 2.7.4

1 0

[alsa-devel] [asoc:for-5.5 32/32] sound/soc/codecs/tas2770.c:211:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 01 Oct '19

01 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: 1a476abc723e644248dba975b71122fcf878703b [32/32] tas2770: add tas2770 smart PA kernel driver reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-37-gd466a02-dirty git checkout 1a476abc723e644248dba975b71122fcf878703b make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2770.c:211:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2770.c:218:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2770.c:225:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer >> sound/soc/codecs/tas2770.c:361:58: sparse: sparse: incorrect type in argument 2 (different base types) @@ expected int bitwidth @@ got restricted snd_pcm_formint bitwidth @@ >> sound/soc/codecs/tas2770.c:361:58: sparse: expected int bitwidth >> sound/soc/codecs/tas2770.c:361:58: sparse: got restricted snd_pcm_format_t vim +211 sound/soc/codecs/tas2770.c 204 205 static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) 206 { 207 int ret; 208 struct snd_soc_component *component = tas2770->component; 209 210 switch (bitwidth) { > 211 case SNDRV_PCM_FORMAT_S16_LE: 212 ret = snd_soc_component_update_bits(component, 213 TAS2770_TDM_CFG_REG2, 214 TAS2770_TDM_CFG_REG2_RXW_MASK, 215 TAS2770_TDM_CFG_REG2_RXW_16BITS); 216 tas2770->v_sense_slot = tas2770->i_sense_slot + 2; 217 break; 218 case SNDRV_PCM_FORMAT_S24_LE: 219 ret = snd_soc_component_update_bits(component, 220 TAS2770_TDM_CFG_REG2, 221 TAS2770_TDM_CFG_REG2_RXW_MASK, 222 TAS2770_TDM_CFG_REG2_RXW_24BITS); 223 tas2770->v_sense_slot = tas2770->i_sense_slot + 4; 224 break; 225 case SNDRV_PCM_FORMAT_S32_LE: 226 ret = snd_soc_component_update_bits(component, 227 TAS2770_TDM_CFG_REG2, 228 TAS2770_TDM_CFG_REG2_RXW_MASK, 229 TAS2770_TDM_CFG_REG2_RXW_32BITS); 230 tas2770->v_sense_slot = tas2770->i_sense_slot + 4; 231 break; 232 233 default: 234 return -EINVAL; 235 } 236 237 tas2770->channel_size = bitwidth; 238 239 ret = snd_soc_component_update_bits(component, 240 TAS2770_TDM_CFG_REG5, 241 TAS2770_TDM_CFG_REG5_VSNS_MASK | 242 TAS2770_TDM_CFG_REG5_50_MASK, 243 TAS2770_TDM_CFG_REG5_VSNS_ENABLE | 244 tas2770->v_sense_slot); 245 if (ret) 246 goto end; 247 ret = snd_soc_component_update_bits(component, 248 TAS2770_TDM_CFG_REG6, 249 TAS2770_TDM_CFG_REG6_ISNS_MASK | 250 TAS2770_TDM_CFG_REG6_50_MASK, 251 TAS2770_TDM_CFG_REG6_ISNS_ENABLE | 252 tas2770->i_sense_slot); 253 254 end: 255 return ret; 256 } 257 258 static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) 259 { 260 int ret; 261 struct snd_soc_component *component = tas2770->component; 262 263 switch (samplerate) { 264 case 48000: 265 ret = snd_soc_component_update_bits(component, 266 TAS2770_TDM_CFG_REG0, 267 TAS2770_TDM_CFG_REG0_SMP_MASK, 268 TAS2770_TDM_CFG_REG0_SMP_48KHZ); 269 if (ret) 270 goto end; 271 ret = snd_soc_component_update_bits(component, 272 TAS2770_TDM_CFG_REG0, 273 TAS2770_TDM_CFG_REG0_31_MASK, 274 TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); 275 if (ret) 276 goto end; 277 break; 278 case 44100: 279 ret = snd_soc_component_update_bits(component, 280 TAS2770_TDM_CFG_REG0, 281 TAS2770_TDM_CFG_REG0_SMP_MASK, 282 TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); 283 if (ret) 284 goto end; 285 ret = snd_soc_component_update_bits(component, 286 TAS2770_TDM_CFG_REG0, 287 TAS2770_TDM_CFG_REG0_31_MASK, 288 TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); 289 if (ret) 290 goto end; 291 break; 292 case 96000: 293 ret = snd_soc_component_update_bits(component, 294 TAS2770_TDM_CFG_REG0, 295 TAS2770_TDM_CFG_REG0_SMP_MASK, 296 TAS2770_TDM_CFG_REG0_SMP_48KHZ); 297 if (ret) 298 goto end; 299 ret = snd_soc_component_update_bits(component, 300 TAS2770_TDM_CFG_REG0, 301 TAS2770_TDM_CFG_REG0_31_MASK, 302 TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); 303 break; 304 case 88200: 305 ret = snd_soc_component_update_bits(component, 306 TAS2770_TDM_CFG_REG0, 307 TAS2770_TDM_CFG_REG0_SMP_MASK, 308 TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); 309 if (ret) 310 goto end; 311 ret = snd_soc_component_update_bits(component, 312 TAS2770_TDM_CFG_REG0, 313 TAS2770_TDM_CFG_REG0_31_MASK, 314 TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); 315 break; 316 case 19200: 317 ret = snd_soc_component_update_bits(component, 318 TAS2770_TDM_CFG_REG0, 319 TAS2770_TDM_CFG_REG0_SMP_MASK, 320 TAS2770_TDM_CFG_REG0_SMP_48KHZ); 321 if (ret) 322 goto end; 323 ret = snd_soc_component_update_bits(component, 324 TAS2770_TDM_CFG_REG0, 325 TAS2770_TDM_CFG_REG0_31_MASK, 326 TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); 327 if (ret) 328 goto end; 329 break; 330 case 17640: 331 ret = snd_soc_component_update_bits(component, 332 TAS2770_TDM_CFG_REG0, 333 TAS2770_TDM_CFG_REG0_SMP_MASK, 334 TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); 335 if (ret) 336 goto end; 337 ret = snd_soc_component_update_bits(component, 338 TAS2770_TDM_CFG_REG0, 339 TAS2770_TDM_CFG_REG0_31_MASK, 340 TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); 341 break; 342 default: 343 ret = -EINVAL; 344 } 345 346 end: 347 if (!ret) 348 tas2770->sampling_rate = samplerate; 349 return ret; 350 } 351 352 static int tas2770_hw_params(struct snd_pcm_substream *substream, 353 struct snd_pcm_hw_params *params, 354 struct snd_soc_dai *dai) 355 { 356 struct snd_soc_component *component = dai->component; 357 struct tas2770_priv *tas2770 = 358 snd_soc_component_get_drvdata(component); 359 int ret; 360 > 361 ret = tas2770_set_bitwidth(tas2770, params_format(params)); 362 if (ret < 0) 363 goto end; 364 365 366 ret = tas2770_set_samplerate(tas2770, params_rate(params)); 367 368 end: 369 return ret; 370 } 371 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:for-5.5 29/32] sound/core/pcm_dmaengine.c:429:50: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 01 Oct '19

01 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: e957204e732bc2916a241dc61dd7dd14e9a98350 [29/32] ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-37-gd466a02-dirty git checkout e957204e732bc2916a241dc61dd7dd14e9a98350 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/core/pcm_dmaengine.c:429:50: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/core/pcm_dmaengine.c:429:55: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/core/pcm_dmaengine.c:429:79: sparse: sparse: restricted snd_pcm_format_t degrades to integer vim +429 sound/core/pcm_dmaengine.c 371 372 /** 373 * snd_dmaengine_pcm_refine_runtime_hwparams - Refine runtime hw params 374 * @substream: PCM substream 375 * @dma_data: DAI DMA data 376 * @hw: PCM hw params 377 * @chan: DMA channel to use for data transfers 378 * 379 * Returns 0 on success, a negative error code otherwise. 380 * 381 * This function will query DMA capability, then refine the pcm hardware 382 * parameters. 383 */ 384 int snd_dmaengine_pcm_refine_runtime_hwparams( 385 struct snd_pcm_substream *substream, 386 struct snd_dmaengine_dai_dma_data *dma_data, 387 struct snd_pcm_hardware *hw, 388 struct dma_chan *chan) 389 { 390 struct dma_slave_caps dma_caps; 391 u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | 392 BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | 393 BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); 394 snd_pcm_format_t i; 395 int ret = 0; 396 397 if (!hw || !chan || !dma_data) 398 return -EINVAL; 399 400 ret = dma_get_slave_caps(chan, &dma_caps); 401 if (ret == 0) { 402 if (dma_caps.cmd_pause && dma_caps.cmd_resume) 403 hw->info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; 404 if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) 405 hw->info |= SNDRV_PCM_INFO_BATCH; 406 407 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 408 addr_widths = dma_caps.dst_addr_widths; 409 else 410 addr_widths = dma_caps.src_addr_widths; 411 } 412 413 /* 414 * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep 415 * hw.formats set to 0, meaning no restrictions are in place. 416 * In this case it's the responsibility of the DAI driver to 417 * provide the supported format information. 418 */ 419 if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) 420 /* 421 * Prepare formats mask for valid/allowed sample types. If the 422 * dma does not have support for the given physical word size, 423 * it needs to be masked out so user space can not use the 424 * format which produces corrupted audio. 425 * In case the dma driver does not implement the slave_caps the 426 * default assumption is that it supports 1, 2 and 4 bytes 427 * widths. 428 */ > 429 for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) { 430 int bits = snd_pcm_format_physical_width(i); 431 432 /* 433 * Enable only samples with DMA supported physical 434 * widths 435 */ 436 switch (bits) { 437 case 8: 438 case 16: 439 case 24: 440 case 32: 441 case 64: 442 if (addr_widths & (1 << (bits / 8))) 443 hw->formats |= pcm_format_to_bits(i); 444 break; 445 default: 446 /* Unsupported types */ 447 break; 448 } 449 } 450 451 return ret; 452 } 453 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_refine_runtime_hwparams); 454 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:for-5.5 2/32] sound/soc/fsl/fsl_mqs.c:182:38: sparse: sparse: Using plain integer as NULL pointer
by kbuild test robot 01 Oct '19

01 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: 9e28f6532c611c0c3fa759d2101aba9f0d41e860 [2/32] ASoC: fsl_mqs: Add MQS component driver reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-37-gd466a02-dirty git checkout 9e28f6532c611c0c3fa759d2101aba9f0d41e860 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/fsl/fsl_mqs.c:182:38: sparse: sparse: Using plain integer as NULL pointer vim +182 sound/soc/fsl/fsl_mqs.c 178 179 static int fsl_mqs_probe(struct platform_device *pdev) 180 { 181 struct device_node *np = pdev->dev.of_node; > 182 struct device_node *gpr_np = 0; 183 struct fsl_mqs *mqs_priv; 184 void __iomem *regs; 185 int ret = 0; 186 187 mqs_priv = devm_kzalloc(&pdev->dev, sizeof(*mqs_priv), GFP_KERNEL); 188 if (!mqs_priv) 189 return -ENOMEM; 190 191 /* On i.MX6sx the MQS control register is in GPR domain 192 * But in i.MX8QM/i.MX8QXP the control register is moved 193 * to its own domain. 194 */ 195 if (of_device_is_compatible(np, "fsl,imx8qm-mqs")) 196 mqs_priv->use_gpr = false; 197 else 198 mqs_priv->use_gpr = true; 199 200 if (mqs_priv->use_gpr) { 201 gpr_np = of_parse_phandle(np, "gpr", 0); 202 if (IS_ERR(gpr_np)) { 203 dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); 204 ret = PTR_ERR(gpr_np); 205 goto out; 206 } 207 208 mqs_priv->regmap = syscon_node_to_regmap(gpr_np); 209 if (IS_ERR(mqs_priv->regmap)) { 210 dev_err(&pdev->dev, "failed to get gpr regmap\n"); 211 ret = PTR_ERR(mqs_priv->regmap); 212 goto out; 213 } 214 } else { 215 regs = devm_platform_ioremap_resource(pdev, 0); 216 if (IS_ERR(regs)) 217 return PTR_ERR(regs); 218 219 mqs_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, 220 "core", 221 regs, 222 &fsl_mqs_regmap_config); 223 if (IS_ERR(mqs_priv->regmap)) { 224 dev_err(&pdev->dev, "failed to init regmap: %ld\n", 225 PTR_ERR(mqs_priv->regmap)); 226 return PTR_ERR(mqs_priv->regmap); 227 } 228 229 mqs_priv->ipg = devm_clk_get(&pdev->dev, "core"); 230 if (IS_ERR(mqs_priv->ipg)) { 231 dev_err(&pdev->dev, "failed to get the clock: %ld\n", 232 PTR_ERR(mqs_priv->ipg)); 233 goto out; 234 } 235 } 236 237 mqs_priv->mclk = devm_clk_get(&pdev->dev, "mclk"); 238 if (IS_ERR(mqs_priv->mclk)) { 239 dev_err(&pdev->dev, "failed to get the clock: %ld\n", 240 PTR_ERR(mqs_priv->mclk)); 241 goto out; 242 } 243 244 dev_set_drvdata(&pdev->dev, mqs_priv); 245 pm_runtime_enable(&pdev->dev); 246 247 return devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, 248 &fsl_mqs_dai, 1); 249 out: 250 if (!IS_ERR(gpr_np)) 251 of_node_put(gpr_np); 252 253 return ret; 254 } 255 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 00/13] ASoC: SOF: fixes and improvements
by Pierre-Louis Bossart 01 Oct '19

01 Oct '19

No new features in this patchset but fixes and improvements identified during validation/product hardening/SoundWire development. I realize we are at the end of the merge window so will let Takashi/Mark decide if this series can be provided as fixes for v5.4 or queued for later. Kai Vehmanen (2): ASoC: SOF: pcm: fix resource leak in hw_free ASoC: SOF: Intel: hda: fix warnings during FW load Keyon Jie (1): ASoC: SOF: topology: fix parse fail issue for byte/bool tuple types Liam Girdwood (3): ASoC: SOF: ipc: retain DSP context after FW exception. ASoC: SOF: Intel: initialise and verify FW crash dump data. ASoC: SOF: Intel: Add context data to any IPC timeout. Marcin Rajwa (2): ASoC: SOF: ipc: introduce no_stream_position in sof_ipc_stream_params struct ASoC: SOF: Intel: hda: fix reset of host_period_bytes Pan Xiuli (1): ASoC: SOF: pcm: harden PCM STOP sequence Pierre-Louis Bossart (2): ASoC: SOF: loader: fix kernel oops on firmware boot failure ASoC: SOF: trace: move to opt-in with Kconfig and module parameter Ranjani Sridharan (2): ASoC: SOF: core: check for mandatory fw_ready op during SOF probe ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture include/sound/sof/stream.h | 4 +-- sound/soc/sof/Kconfig | 16 ++++++++++++ sound/soc/sof/core.c | 28 +++++++++++++++----- sound/soc/sof/debug.c | 16 ++++++++++++ sound/soc/sof/intel/Kconfig | 10 +++++++ sound/soc/sof/intel/bdw.c | 29 +++++++++++++++++++- sound/soc/sof/intel/byt.c | 29 +++++++++++++++++++- sound/soc/sof/intel/hda-ctrl.c | 12 +++------ sound/soc/sof/intel/hda-loader.c | 1 + sound/soc/sof/intel/hda-pcm.c | 15 ++++++++--- sound/soc/sof/intel/hda-stream.c | 45 +++++++++++++++++++++++++++----- sound/soc/sof/intel/hda.c | 7 +++++ sound/soc/sof/intel/hda.h | 5 +++- sound/soc/sof/ipc.c | 10 +------ sound/soc/sof/loader.c | 4 +-- sound/soc/sof/pcm.c | 35 +++++++++++++++++-------- sound/soc/sof/sof-priv.h | 17 ++++++++---- sound/soc/sof/topology.c | 4 ++- sound/soc/sof/trace.c | 17 +++++++++++- 19 files changed, 245 insertions(+), 59 deletions(-) -- 2.20.1

2 26

[alsa-devel] [PATCH] ASoC: wm_adsp: Fix theoretical NULL pointer for alg_region
by Charles Keepax 01 Oct '19

01 Oct '19

From: Li Xu <li.xu(a)cirrus.com> Fix potential NULL pointer dereference for alg_region in wm_adsp_buffer_parse_legacy. In practice this can never happen as loading the firmware should have failed at the wm_adsp2_setup_algs stage, however probably better for the code to be robust against future changes and this is more helpful for static analysis. Signed-off-by: Li Xu <li.xu(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ae28d9907c30f..85396d920e0a9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3697,11 +3697,16 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp) u32 xmalg, addr, magic; int i, ret; + alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); + if (!alg_region) { + adsp_err(dsp, "No algorithm region found\n"); + return -EINVAL; + } + buf = wm_adsp_buffer_alloc(dsp); if (!buf) return -ENOMEM; - alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); xmalg = dsp->ops->sys_config_size / sizeof(__be32); addr = alg_region->base + xmalg + ALG_XM_FIELD(magic); -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: madera: Add support for using MCLK3
by Charles Keepax 01 Oct '19

01 Oct '19

Some Madera CODECs have a third MCLK pin allow this to be sourced for the FLLs and system clocks. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/madera.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/madera.h b/sound/soc/codecs/madera.h index 1f3e8e230cf25..6d8938a3fb647 100644 --- a/sound/soc/codecs/madera.h +++ b/sound/soc/codecs/madera.h @@ -27,6 +27,7 @@ #define MADERA_FLL_SRC_NONE -1 #define MADERA_FLL_SRC_MCLK1 0 #define MADERA_FLL_SRC_MCLK2 1 +#define MADERA_FLL_SRC_MCLK3 2 #define MADERA_FLL_SRC_SLIMCLK 3 #define MADERA_FLL_SRC_FLL1 4 #define MADERA_FLL_SRC_FLL2 5 @@ -51,6 +52,7 @@ #define MADERA_CLK_SRC_MCLK1 0x0 #define MADERA_CLK_SRC_MCLK2 0x1 +#define MADERA_CLK_SRC_MCLK3 0x2 #define MADERA_CLK_SRC_FLL1 0x4 #define MADERA_CLK_SRC_FLL2 0x5 #define MADERA_CLK_SRC_FLL3 0x6 -- 2.11.0

2 1

[alsa-devel] [PATCH 0/4] ASoC: Intel: boards: improvements for RT5651/82-based platforms
by Pierre-Louis Bossart 01 Oct '19

01 Oct '19

RT5651/RT5682 boards are used for SOF validation. This patchset corrects problems identified during module load/reload tests and an error for DMIC support in some configurations. These fixes are not critical and can be queued for v5.5 Bard Liao (1): ASoC: intel: sof_rt5682: use separate route map for dmic Jaska Uimonen (3): ASoC: intel: bytcr_rt5651: add null check to support_button_press ASoC: rt5682: add NULL handler to set_jack function ASoC: intel: sof_rt5682: add remove function to disable jack sound/soc/codecs/rt5651.c | 3 ++ sound/soc/codecs/rt5682.c | 12 +++++-- sound/soc/intel/boards/sof_rt5682.c | 51 ++++++++++++++++++++++++++--- 3 files changed, 60 insertions(+), 6 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH] ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency
by Arnd Bergmann 01 Oct '19

01 Oct '19

CONFIG_SND_SOC_SOF_IMX depends on CONFIG_SND_SOC_SOF, but is in turn referenced by the sof-of-dev driver. This creates a reverse dependency that manifests in a link error when CONFIG_SND_SOC_SOF_OF is built-in but CONFIG_SND_SOC_SOF_IMX=m: sound/soc/sof/sof-of-dev.o:(.data+0x118): undefined reference to `sof_imx8_ops' Make the latter a 'bool' symbol and change the Makefile so the imx8 driver is compiled the same way as the driver using it. A nicer way would be to reverse the layering and move all the imx specific bits of sof-of-dev.c into the imx driver itself, which can then call into the common code. Doing this would need more testing and can be done if we add another driver like the first one. Fixes: f4df4e4042b0 ("ASoC: SOF: imx8: Fix COMPILE_TEST error") Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sof/imx/Kconfig | 2 +- sound/soc/sof/imx/Makefile | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750..a3891654a1fc 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -12,7 +12,7 @@ config SND_SOC_SOF_IMX_TOPLEVEL if SND_SOC_SOF_IMX_TOPLEVEL config SND_SOC_SOF_IMX8 - tristate "SOF support for i.MX8" + bool "SOF support for i.MX8" depends on IMX_SCU depends on IMX_DSP help diff --git a/sound/soc/sof/imx/Makefile b/sound/soc/sof/imx/Makefile index 6ef908e8c807..9e8f35df0ff2 100644 --- a/sound/soc/sof/imx/Makefile +++ b/sound/soc/sof/imx/Makefile @@ -1,4 +1,6 @@ # SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) snd-sof-imx8-objs := imx8.o -obj-$(CONFIG_SND_SOC_SOF_IMX8) += snd-sof-imx8.o +ifdef CONFIG_SND_SOC_SOF_IMX8 +obj-$(CONFIG_SND_SOC_SOF_OF) += snd-sof-imx8.o +endif -- 2.20.0

3 5

[alsa-devel] [PATCH] ASoC: max98373: check for device node before parsing
by Sathyanarayana Nujella 01 Oct '19

01 Oct '19

Below Oops is caused in a system which uses ACPI instead of device node: of_get_named_gpiod_flags: can't parse 'maxim,reset-gpio' property of node '(null)[0]' BUG: kernel NULL pointer dereference, address: 0000000000000010 This patch avoids NULL pointer deferencing by adding a check before parsing and initializes to make reset-gpio pin as invalid. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> --- sound/soc/codecs/max98373.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index e609abcf3220..eb709d528259 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -901,16 +901,20 @@ static void max98373_slot_config(struct i2c_client *i2c, max98373->i_slot = value & 0xF; else max98373->i_slot = 1; - - max98373->reset_gpio = of_get_named_gpio(dev->of_node, + if (dev->of_node) { + max98373->reset_gpio = of_get_named_gpio(dev->of_node, "maxim,reset-gpio", 0); - if (!gpio_is_valid(max98373->reset_gpio)) { - dev_err(dev, "Looking up %s property in node %s failed %d\n", - "maxim,reset-gpio", dev->of_node->full_name, - max98373->reset_gpio); + if (!gpio_is_valid(max98373->reset_gpio)) { + dev_err(dev, "Looking up %s property in node %s failed %d\n", + "maxim,reset-gpio", dev->of_node->full_name, + max98373->reset_gpio); + } else { + dev_dbg(dev, "maxim,reset-gpio=%d", + max98373->reset_gpio); + } } else { - dev_dbg(dev, "maxim,reset-gpio=%d", - max98373->reset_gpio); + /* this makes reset_gpio as invalid */ + max98373->reset_gpio = -1; } if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value)) -- 2.7.4

2 1

[alsa-devel] [PATCH V2 1/2] ASoC: fsl_mqs: add DT binding documentation
by Shengjiu Wang 01 Oct '19

01 Oct '19

Add the DT binding documentation for NXP MQS driver Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 -refine the comments for properties .../devicetree/bindings/sound/fsl,mqs.txt | 36 +++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt new file mode 100644 index 000000000000..40353fc30255 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -0,0 +1,36 @@ +fsl,mqs audio CODEC + +Required properties: + - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + - clocks : A list of phandles + clock-specifiers, one for each entry in + clock-names + - clock-names : "mclk" - must required. + "core" - required if compatible is "fsl,imx8qm-mqs", it + is for register access. + - gpr : A phandle of General Purpose Registers in IOMUX Controller. + Required if compatible is "fsl,imx6sx-mqs". + +Required if compatible is "fsl,imx8qm-mqs": + - power-domains: A phandle of PM domain provider node. + - reg: Offset and length of the register set for the device. + +Example: + +mqs: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + status = "disabled"; +}; + +mqs: mqs@59850000 { + compatible = "fsl,imx8qm-mqs"; + reg = <0x59850000 0x10000>; + clocks = <&clk IMX8QM_AUD_MQS_IPG>, + <&clk IMX8QM_AUD_MQS_HMCLK>; + clock-names = "core", "mclk"; + power-domains = <&pd_mqs0>; + status = "disabled"; +}; -- 2.21.0

4 6

[alsa-devel] [PATCH v2] ASoC: Intel: cht_bsw_rt5645: Add quirk for boards using pmc_plt_clk_0
by Sam McNally 01 Oct '19

01 Oct '19

As of commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL"), the cht_bsw_rt5645 driver needs to enable the clock it's using for the codec's mclk. It does this from commit 7735bce05a9c ("ASoC: Intel: boards: use devm_clk_get() unconditionally"), enabling pmc_plt_clk_3. However, Strago family Chromebooks use pmc_plt_clk_0 for the codec mclk, resulting in white noise with some digital microphones. Add a DMI-based quirk for Strago family Chromebooks to use pmc_plt_clk_0 instead - mirroring the changes made to cht_bsw_max98090_ti in commit a182ecd3809c ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0") and making use of the existing dmi_check_system() call and related infrastructure added in commit 22af29114eb4 ("ASoC: Intel: cht-bsw-rt5645: add quirks for SSP0/AIF1/AIF2 routing"). Signed-off-by: Sam McNally <sammc(a)chromium.org> --- Changes in v2: - Added to the description references to more related commits (similar cht_bsw_max98090_ti clock quirks inspiring this change and and the cht_bsw_rt5645 quirks setup) sound/soc/intel/boards/cht_bsw_rt5645.c | 26 +++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 8879c3be29d5..c68a5b85a4a0 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -48,6 +48,7 @@ struct cht_mc_private { #define CHT_RT5645_SSP2_AIF2 BIT(16) /* default is using AIF1 */ #define CHT_RT5645_SSP0_AIF1 BIT(17) #define CHT_RT5645_SSP0_AIF2 BIT(18) +#define CHT_RT5645_PMC_PLT_CLK_0 BIT(19) static unsigned long cht_rt5645_quirk = 0; @@ -59,6 +60,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled"); if (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + dev_info(dev, "quirk PMC_PLT_CLK_0 enabled"); } static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -226,15 +229,21 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, return 0; } -/* uncomment when we have a real quirk static int cht_rt5645_quirk_cb(const struct dmi_system_id *id) { cht_rt5645_quirk = (unsigned long)id->driver_data; return 1; } -*/ static const struct dmi_system_id cht_rt5645_quirk_table[] = { + { + /* Strago family Chromebooks */ + .callback = cht_rt5645_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_Strago"), + }, + .driver_data = (void *)CHT_RT5645_PMC_PLT_CLK_0, + }, { }, }; @@ -526,6 +535,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int dai_index = 0; int ret_val = 0; int i; + const char *mclk_name; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -662,11 +672,15 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (ret_val) return ret_val; - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + dev_err(&pdev->dev, "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.23.0.237.gc6a4ce50a0-goog

3 2

[alsa-devel] [PATCH v3 1/2] ASoC: dt-bindings: Convert Allwinner A10 codec to a schema
by Maxime Ripard 01 Oct '19

01 Oct '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> The Allwinner SoCs have an embedded audio codec that is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v2: - Change the audio-routing values to an enum, and enforce boundaries on the size - Add restrictions to the possible values of audio-routing Changes from v1: - Fix subject prefix --- .../sound/allwinner,sun4i-a10-codec.yaml | 262 ++++++++++++++++++ .../devicetree/bindings/sound/sun4i-codec.txt | 94 ------- 2 files changed, 262 insertions(+), 94 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000000..faa75b91c072 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,262 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + 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. + allOf: + - $ref: /schemas/types.yaml#definitions/non-unique-string-array + - minItems: 2 + maxItems: 18 + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt deleted file mode 100644 index 66579bbd3294..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ /dev/null @@ -1,94 +0,0 @@ -* Allwinner A10 Codec - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun4i-a10-codec" - - "allwinner,sun6i-a31-codec" - - "allwinner,sun7i-a20-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- dmas: DMA channels for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain the following: - - "apb": the parent APB clock for this controller - - "codec": the parent module clock - -Optional properties: -- allwinner,pa-gpios: gpio to enable external amplifier - -Required properties for the following compatibles: - - "allwinner,sun6i-a31-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- resets: phandle to the reset control for this device -- allwinner,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 include: - - Audio pins on the SoC: - "HP" - "HPCOM" - "LINEIN" (not on sun8i-v3s) - "LINEOUT" (not on sun8i-a23 or sun8i-v3s) - "MIC1" - "MIC2" (not on sun8i-v3s) - "MIC3" (sun6i-a31 only) - - Microphone biases from the SoC: - "HBIAS" - "MBIAS" (not on sun8i-v3s) - - Board connectors: - "Headphone" - "Headset Mic" - "Line In" - "Line Out" - "Mic" - "Speaker" - -Required properties for the following compatibles: - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- allwinner,codec-analog-controls: A phandle to the codec analog controls - block in the PRCM. - -Example: -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun7i-a20-codec"; - reg = <0x01c22c00 0x40>; - interrupts = <0 30 4>; - clocks = <&apb0_gates 0>, <&codec_clk>; - clock-names = "apb", "codec"; - dmas = <&dma 0 19>, <&dma 0 19>; - dma-names = "rx", "tx"; -}; - -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun6i-a31-codec"; - reg = <0x01c22c00 0x98>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; - clock-names = "apb", "codec"; - resets = <&ccu RST_APB1_CODEC>; - dmas = <&dma 15>, <&dma 15>; - dma-names = "rx", "tx"; - allwinner,audio-routing = - "Headphone", "HP", - "Speaker", "LINEOUT", - "LINEIN", "Line In", - "MIC1", "MBIAS", - "MIC1", "Mic", - "MIC2", "HBIAS", - "MIC2", "Headset Mic"; -}; -- 2.21.0

4 4

[alsa-devel] [PATCH] ASoC: xlnx: Use devm_platform_ioremap_resource() in xlnx_formatter_pcm_probe()
by Markus Elfring 01 Oct '19

01 Oct '19

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 19 Sep 2019 17:27:57 +0200 Simplify this function implementation by using a known wrapper function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/xilinx/xlnx_formatter_pcm.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 48970efe7838..fb652e73abeb 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -564,7 +564,6 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) int ret; u32 val; struct xlnx_pcm_drv_data *aud_drv_data; - struct resource *res; struct device *dev = &pdev->dev; aud_drv_data = devm_kzalloc(dev, sizeof(*aud_drv_data), GFP_KERNEL); @@ -584,13 +583,7 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "audio formatter node:addr to resource failed\n"); - ret = -ENXIO; - goto clk_err; - } - aud_drv_data->mmio = devm_ioremap_resource(dev, res); + aud_drv_data->mmio = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(aud_drv_data->mmio)) { dev_err(dev, "audio formatter ioremap failed\n"); ret = PTR_ERR(aud_drv_data->mmio); -- 2.23.0

2 1

[alsa-devel] [PATCH v2] ASoC: soc-component.h: remove GPL explanation from header
by Kuninori Morimoto 01 Oct '19

01 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-component.h already has SPDX License, thus, GPL explanation is not needed. This patch removes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark - this is v2 of "[PATCH] ASoC: soc-component.h: tidyup for soc-component.h" - Subject was renamed - keep "soc.h" - do nothing to header-check include/sound/soc-component.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2e..1a832695 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -3,10 +3,6 @@ * soc-component.h * * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SOC_COMPONENT_H #define __SOC_COMPONENT_H -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: dt-bindings: mt8183: add missing update
by Tzung-Bi Shih 01 Oct '19

01 Oct '19

Headset codec is optional. Add missing update to DT binding document. Fixes: a962a809e5e4 ("ASoC: mediatek: mt8183: make headset codec optional") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../bindings/sound/mt8183-mt6358-ts3a227-max98357.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt index d6d5207fa996..17ff3892f439 100644 --- a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt +++ b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt @@ -2,9 +2,11 @@ MT8183 with MT6358, TS3A227 and MAX98357 CODECS Required properties: - compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" -- mediatek,headset-codec: the phandles of ts3a227 codecs - mediatek,platform: the phandle of MT8183 ASoC platform +Optional properties: +- mediatek,headset-codec: the phandles of ts3a227 codecs + Example: sound { -- 2.23.0.351.gc4317032e6-goog

2 1

[alsa-devel] [PATCH v2 00/14] ASoC: soc-core cleanup - step 3
by Kuninori Morimoto 01 Oct '19

01 Oct '19

Hi Mark These are v2 of soc-core.c cleanup (maybe) step 3. These try to clenaup around kmalloc/kfree related code. Basically these do nothing from technical point of view. I added +1 extra patch as [00/14] Kuninori Morimoto (14): ASoC: soc-core: move soc_free_pcm_runtime() ASoC: soc-core: merge soc_add_pcm_runtime() into soc_new_pcm_runtime() ASoC: soc-core: call list_del(&rtd->list) at soc_free_pcm_runtime() ASoC: soc-core: create rtd->codec_dais first ASoC: soc-core: merge soc_new_pcm_runtime() and soc_rtd_init() ASoC: soc-core: merge soc_free_pcm_runtime() and soc_rtd_free() ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order ASoC: soc-core: remove snd_soc_rtdcom_del_all() ASoC: soc-core: use devm_kzalloc() for rtd ASoC: soc-core: remove soc_remove_dai_links() ASoC: soc-core: add soc_setup_card_name() ASoC: soc-core: use devm_xxx for component related resource ASoC: soc-core: setup card->rtd_num at snd_soc_instantiate_card() ASoC: soc-ops: use snd_soc_card_get_kcontrol() at snd_soc_limit_volume() include/sound/soc.h | 1 - sound/soc/soc-core.c | 287 ++++++++++++++++++++++++++------------------------- sound/soc/soc-ops.c | 11 +- 3 files changed, 151 insertions(+), 148 deletions(-) -- 2.7.4

2 22

[alsa-devel] [PATCH v7] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 01 Oct '19

01 Oct '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: amd: acp3x: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index bc4dfafdfcd1..ea57088d50ce 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -631,7 +631,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); - return -ENODEV; + return -ENODEV; } adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: rt5663: clean up indentation issues
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There are two break statements that are indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5663.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 2943692f66ed..e6c1ec6c426e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3644,7 +3644,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_1, RT5663_LDO1_DVO_MASK | RT5663_AMP_HP_MASK, RT5663_LDO1_DVO_0_9V | RT5663_AMP_HP_3X); - break; + break; case CODEC_VER_0: regmap_update_bits(rt5663->regmap, RT5663_DIG_MISC, RT5663_DIG_GATE_CTRL_MASK, RT5663_DIG_GATE_CTRL_EN); @@ -3663,7 +3663,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_TDM_2, RT5663_DATA_SWAP_ADCDAT1_MASK, RT5663_DATA_SWAP_ADCDAT1_LL); - break; + break; default: dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); } -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: wcd9335: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is an if statement that is indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wcd9335.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f318403133e9..f11ffa28683b 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2837,11 +2837,11 @@ static int wcd9335_codec_enable_dec(struct snd_soc_dapm_widget *w, TX_HPF_CUT_OFF_FREQ_MASK) >> 5; snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x10); snd_soc_component_update_bits(comp, dec_cfg_reg, 0x08, 0x00); - if (hpf_coff_freq != CF_MIN_3DB_150HZ) { - snd_soc_component_update_bits(comp, dec_cfg_reg, - TX_HPF_CUT_OFF_FREQ_MASK, - hpf_coff_freq << 5); - } + if (hpf_coff_freq != CF_MIN_3DB_150HZ) { + snd_soc_component_update_bits(comp, dec_cfg_reg, + TX_HPF_CUT_OFF_FREQ_MASK, + hpf_coff_freq << 5); + } break; case SND_SOC_DAPM_POST_PMD: snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x00); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: stm32: sai: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/stm/stm32_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index ef4273361d0d..e20267504b16 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -100,7 +100,7 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); - stm32_sai_pclk_disable(&sai->pdev->dev); + stm32_sai_pclk_disable(&sai->pdev->dev); return -EINVAL; } -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: core: Clarify usage of ignore_machine
by Daniel Baluta 01 Oct '19

01 Oct '19

For a sound card ignore_machine means that existing FEs links should be ignored and existing BEs links should be overridden with some information from the matching component driver. Current code make some confusions about this so fix it! Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 88978a3036c4..e32a45f6bd88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1853,7 +1853,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) for_each_component(component) { - /* does this component override FEs ? */ + /* does this component override BEs ? */ if (!component->driver->ignore_machine) continue; @@ -1874,7 +1874,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) continue; } - dev_info(card->dev, "info: override FE DAI link %s\n", + dev_info(card->dev, "info: override BE DAI link %s\n", card->dai_link[i].name); /* override platform component */ -- 2.17.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

2 7

[alsa-devel] [PATCH v2] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 01 Oct '19

01 Oct '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Rebased on top 'ASoC: soc-component: fix a couple missing error assignments' Changes since rfc: - Fixed missing error assignments (copy pasted the error) - Document the trigger order in the commit message Regards, Peter sound/soc/soc-pcm.c | 64 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a1b99ac57d9e..66910500e3b6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1074,6 +1074,42 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 2

[alsa-devel] [PATCH] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 01 Oct '19

01 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index aa7efd4da1c8..c60e951122c9 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -1982,6 +1982,15 @@ static void hdmi_disable_overflow_interrupts(struct dw_hdmi *hdmi) HDMI_IH_MUTE_FC_STAT2); } +static void dw_hdmi_audio_restore(struct dw_hdmi *hdmi) +{ + mutex_lock(&hdmi->audio_mutex); + + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); + + mutex_unlock(&hdmi->audio_mutex); +} + static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) { int ret; @@ -2045,7 +2054,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + dw_hdmi_audio_restore(hdmi); } /* not for DVI mode */ -- 2.23.0.444.g18eeb5a265-goog

1 0

[alsa-devel] [PATCH 1/5] ASoC: amd: Registering device endpoints using MFD framework
by Ravulapati Vishnu vardhan rao 30 Sep '19

30 Sep '19

Removed platform based endpoint registering. Now Registering DMA and multiple I2S BT playback endpoint devices automatically by using MFD framework. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Reviewed-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 54 ++++++++------- sound/soc/amd/raven/pci-acp3x.c | 148 +++++++++++++++++++++++++++++----------- 2 files changed, 139 insertions(+), 63 deletions(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 4f2cadd..3a27fc8 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -7,33 +7,41 @@ #include "chip_offset_byte.h" -#define ACP3x_PHY_BASE_ADDRESS 0x1240000 -#define ACP3x_I2S_MODE 0 -#define ACP3x_REG_START 0x1240000 -#define ACP3x_REG_END 0x1250200 -#define I2S_MODE 0x04 -#define BT_TX_THRESHOLD 26 -#define BT_RX_THRESHOLD 25 -#define ACP3x_POWER_ON 0x00 -#define ACP3x_POWER_ON_IN_PROGRESS 0x01 -#define ACP3x_POWER_OFF 0x02 -#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_DEVS 3 +#define ACP3x_PHY_BASE_ADDRESS 0x1240000 +#define ACP3x_I2S_MODE 0 +#define ACP3x_REG_START 0x1240000 +#define ACP3x_REG_END 0x1250200 +#define ACP3x_I2STDM_REG_START 0x1242400 +#define ACP3x_I2STDM_REG_END 0x1242410 +#define ACP3x_BT_TDM_REG_START 0x1242800 +#define ACP3x_BT_TDM_REG_END 0x1242810 +#define I2S_MODE 0x04 +#define I2S_RX_THRESHOLD 27 +#define I2S_TX_THRESHOLD 28 +#define BT_TX_THRESHOLD 26 +#define BT_RX_THRESHOLD 25 +#define ACP_ERR_INTR_MASK 29 +#define ACP3x_POWER_ON 0x00 +#define ACP3x_POWER_ON_IN_PROGRESS 0x01 +#define ACP3x_POWER_OFF 0x02 +#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 #define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 #define ACP_SRAM_PTE_OFFSET 0x02050000 -#define PAGE_SIZE_4K_ENABLE 0x2 +#define PAGE_SIZE_4K_ENABLE 0x2 #define MEM_WINDOW_START 0x4000000 -#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 -#define CAPTURE_FIFO_ADDR_OFFSET 0x500 - -#define PLAYBACK_MIN_NUM_PERIODS 2 -#define PLAYBACK_MAX_NUM_PERIODS 8 -#define PLAYBACK_MAX_PERIOD_SIZE 16384 -#define PLAYBACK_MIN_PERIOD_SIZE 4096 -#define CAPTURE_MIN_NUM_PERIODS 2 -#define CAPTURE_MAX_NUM_PERIODS 8 -#define CAPTURE_MAX_PERIOD_SIZE 16384 -#define CAPTURE_MIN_PERIOD_SIZE 4096 +#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 +#define CAPTURE_FIFO_ADDR_OFFSET 0x500 + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 8 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 8 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 4096 #define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec24..c0b8fad 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -9,23 +9,49 @@ #include <linux/io.h> #include <linux/platform_device.h> #include <linux/interrupt.h> +#include <linux/mfd/core.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> #include "acp3x.h" +struct i2s_platform_data { + unsigned int cap; + int channel; + u32 snd_rates; +}; struct acp3x_dev_data { + struct device *parent; + struct mfd_cell *cell; + struct resource *res; void __iomem *acp3x_base; bool acp3x_audio_mode; - struct resource *res; struct platform_device *pdev; }; + +static struct device *get_mfd_cell_dev(const char *device_name, int r) +{ + char auto_dev_name[25]; + struct device *dev; + + snprintf(auto_dev_name, sizeof(auto_dev_name), + "%s.%d.auto", device_name, r); + dev = bus_find_device_by_name(&platform_bus_type, + NULL, auto_dev_name); + dev_info(dev, "device %s added\n", auto_dev_name); + return dev; +} + static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { int ret; - u32 addr, val; + uint64_t addr; + int val, i, r; struct acp3x_dev_data *adata; - struct platform_device_info pdevinfo; + struct device *dev; + struct i2s_platform_data *i2s_pdata; unsigned int irqflags; if (pci_enable_device(pci)) { @@ -63,47 +89,82 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); - + adata->parent = &pci->dev; val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); switch (val) { case I2S_MODE: adata->res = devm_kzalloc(&pci->dev, - sizeof(struct resource) * 2, - GFP_KERNEL); - if (!adata->res) { + sizeof(struct resource) * 4, + GFP_KERNEL); + adata->cell = devm_kzalloc(&pci->dev, + sizeof(struct mfd_cell) * 3, + GFP_KERNEL); + if (!adata->cell) { ret = -ENOMEM; goto unmap_mmio; } - adata->res[0].name = "acp3x_i2s_iomem"; - adata->res[0].flags = IORESOURCE_MEM; - adata->res[0].start = addr; - adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); - - adata->res[1].name = "acp3x_i2s_irq"; - adata->res[1].flags = IORESOURCE_IRQ; - adata->res[1].start = pci->irq; - adata->res[1].end = pci->irq; - - adata->acp3x_audio_mode = ACP3x_I2S_MODE; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - pdevinfo.name = "acp3x_rv_i2s"; - pdevinfo.id = 0; - pdevinfo.parent = &pci->dev; - pdevinfo.num_res = 2; - pdevinfo.res = adata->res; - pdevinfo.data = &irqflags; - pdevinfo.size_data = sizeof(irqflags); - - adata->pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(adata->pdev)) { - dev_err(&pci->dev, "cannot register %s device\n", - pdevinfo.name); - ret = PTR_ERR(adata->pdev); - goto unmap_mmio; + i2s_pdata = devm_kzalloc(&pci->dev, + sizeof(struct i2s_platform_data) * 3, + GFP_KERNEL); + if (i2s_pdata == NULL) { + kfree(adata->res); + kfree(adata->cell); + return -ENOMEM; } + adata->res[0].name = "acp3x_i2s_iomem"; + adata->res[0].flags = IORESOURCE_MEM; + adata->res[0].start = addr; + adata->res[0].end = addr + + (ACP3x_REG_END - ACP3x_REG_START); + i2s_pdata[0].cap = 0; + i2s_pdata[0].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[1].name = "acp3x_i2s_sp_play_cap"; + adata->res[1].flags = IORESOURCE_MEM; + adata->res[1].start = addr + ACP3x_I2STDM_REG_START; + adata->res[1].end = addr + ACP3x_I2STDM_REG_END; + i2s_pdata[1].cap = 0; + i2s_pdata[1].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[2].name = "acp3x_i2s_bt_play_cap"; + adata->res[2].flags = IORESOURCE_MEM; + adata->res[2].start = addr + ACP3x_BT_TDM_REG_START; + adata->res[2].end = addr + ACP3x_BT_TDM_REG_END; + i2s_pdata[2].cap = 0; + i2s_pdata[2].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[3].name = "acp3x_i2s_irq"; + adata->res[3].flags = IORESOURCE_IRQ; + adata->res[3].start = pci->irq; + adata->res[3].end = adata->res[3].start; + + adata->acp3x_audio_mode = ACP3x_I2S_MODE; + + adata->cell[0].name = "acp3x_rv_i2s_dma"; + adata->cell[0].num_resources = 4; + adata->cell[0].resources = &adata->res[0]; + adata->cell[0].platform_data = &irqflags; + adata->cell[0].pdata_size = sizeof(irqflags); + + adata->cell[1].name = "acp3x_i2s_playcap"; + adata->cell[1].num_resources = 1; + adata->cell[1].resources = &adata->res[1]; + adata->cell[1].platform_data = &i2s_pdata[0]; + adata->cell[1].pdata_size = + sizeof(struct i2s_platform_data); + + adata->cell[2].name = "acp3x_i2s_playcap"; + adata->cell[2].num_resources = 1; + adata->cell[2].resources = &adata->res[2]; + adata->cell[2].platform_data = &i2s_pdata[1]; + adata->cell[2].pdata_size = + sizeof(struct i2s_platform_data); + r = mfd_add_hotplug_devices(adata->parent, adata->cell, 3); + for (i = 0; i < 3 ; i++) + dev = get_mfd_cell_dev(adata->cell[i].name, i); break; + default: dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); ret = -ENODEV; @@ -112,23 +173,29 @@ static int snd_acp3x_probe(struct pci_dev *pci, return 0; unmap_mmio: - pci_disable_msi(pci); + mfd_remove_devices(adata->parent); + kfree(adata->res); + kfree(adata->cell); iounmap(adata->acp3x_base); + /*ignore device status and return driver probe error*/ + return -ENODEV; release_regions: pci_release_regions(pci); + /*ignore device status and return driver probe error*/ + return -ENODEV; disable_pci: pci_disable_device(pci); - - return ret; + /*ignore device status and return driver probe error*/ + return -ENODEV; } static void snd_acp3x_remove(struct pci_dev *pci) { struct acp3x_dev_data *adata = pci_get_drvdata(pci); - - platform_device_unregister(adata->pdev); + mfd_remove_devices(adata->parent); + kfree(adata->res); + kfree(adata->cell); iounmap(adata->acp3x_base); - pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); @@ -151,6 +218,7 @@ static struct pci_driver acp3x_driver = { module_pci_driver(acp3x_driver); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); MODULE_DESCRIPTION("AMD ACP3x PCI driver"); MODULE_LICENSE("GPL v2"); -- 2.7.4

4 9

Re: [alsa-devel] [PATCH v2] ASoC: Intel: Skylake: prevent memory leak in snd_skl_parse_uuids
by Pierre-Louis Bossart 28 Sep '19

28 Sep '19

On 9/26/19 9:55 PM, Navid Emamdoost wrote: > On Wed, Sep 25, 2019 at 12:05:28PM -0500, Pierre-Louis Bossart wrote: >> On 9/25/19 11:19 AM, Navid Emamdoost wrote: >>> In snd_skl_parse_uuids if allocation for module->instance_id fails, the >>> allocated memory for module shoulde be released. I changes the >>> allocation for module to use devm_kzalloc to be resource_managed >>> allocation and avoid the release in error path. >> >> if you use devm_, don't you need to fix the error path as well then, I see a >> kfree(uuid) in skl_freeup_uuid_list(). >> >> I am not very familiar with this code but the error seems to be that the >> list_add_tail() is called after the module->instance_id is allocated, so >> there is a risk that the module allocated earlier is not freed (since it's >> not yet added to the list). Freeing the module as done in patch 1 works, >> using devm_ without fixing the error path does not seem correct to me. >> > Thanks for the feedback, then it's your call if you can accept patch 1 as > fix. Cezary, it's really your call. > Navid. >>> >>> Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> >>> --- >>> Changes in v2: >>> - Changed the allocation for module from kzalloc to devm_kzalloc >>> --- >>> sound/soc/intel/skylake/skl-sst-utils.c | 2 +- >>> 1 file changed, 1 insertion(+), 1 deletion(-) >>> >>> diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c >>> index d43cbf4a71ef..ac37f04b0eea 100644 >>> --- a/sound/soc/intel/skylake/skl-sst-utils.c >>> +++ b/sound/soc/intel/skylake/skl-sst-utils.c >>> @@ -284,7 +284,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, >>> */ >>> for (i = 0; i < num_entry; i++, mod_entry++) { >>> - module = kzalloc(sizeof(*module), GFP_KERNEL); >>> + module = devm_kzalloc(ctx->dev, sizeof(*module), GFP_KERNEL); >>> if (!module) { >>> ret = -ENOMEM; >>> goto free_uuid_list; >>> >>

4 5

[alsa-devel] [PATCH v5] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Marek Szyprowski 27 Sep '19

27 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung I2S controller to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v5: - Removed '#clock-cells' property from required properties - Added deprecated property to 'clock-output-names' and removed corresponding comment. Best regards, Maciej Falkowski --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ----------- .../bindings/sound/samsung-i2s.yaml | 136 ++++++++++++++++++ 2 files changed, 136 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..b9502b82fa9d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,136 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with + secondary fifo, s/w reset control and internal mux for root clk src. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for + playback, stereo channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + + samsung,exynos7-i2s: with all the available features of exynos5 i2s. + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for i2s0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for i2s1 and i2s2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clk. i2s0 has internal mux to select the source + of root clk and i2s1 and i2s2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: + deprecated: true + oneOf: + - items: # for i2s0 + - const: i2s_cdclk0 + - items: # for i2s1 + - const: i2s_cdclk1 + - items: # for i2s2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + sub system(used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; + -- 2.17.1

4 3

[alsa-devel] [PATCH] ASoC: Skylake SST driver - blacklist the PCI device IDs for the auto probe
by Jaroslav Kysela 27 Sep '19

27 Sep '19

There are basically three drivers for the PCI devices for the recent Intel hardware with the build-in DSPs. The legacy HDA driver has dmic_detect module option for the auto detection of the platforms with the digital microphone. Because the SOF driver is preferred, just skip PCI probe in the Skylake SST driver when the PCI device ID clashes by default. The user can override the auto behaviour with the pci_binding module parameter. Boot log from Lenovo Carbon X1 (7th gen) with the default settings: snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, aborting probe snd_soc_skl 0000:00:1f.3: SOF driver is preferred on this platform, aborting probe sof-audio-pci 0000:00:1f.3: warning: No matching ASoC machine driver found sof-audio-pci 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040380 .... Perhaps, it may be more wise to create one shared module and all three drivers should call the driver detection routine(s) from one place. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/skylake/skl.c | 37 +++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..cace55ca8d55 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -38,6 +38,39 @@ static int skl_pci_binding; module_param_named(pci_binding, skl_pci_binding, int, 0444); MODULE_PARM_DESC(pci_binding, "PCI binding (0=auto, 1=only legacy, 2=only asoc"); +/* + * + */ +static int skl_sof_support(struct pci_dev *pci) +{ + /* the SOF driver has same PCI IDs */ + if (pci->vendor == 0x8086) { + switch (pci->device) { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + case 0x02c8: /* CML-LP */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + case 0x06c8: /* CML-H */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + case 0x3198: /* GLK */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + case 0x5a98: /* BXT-P */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + case 0x9dc8: /* CNL */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + case 0xa348: /* CFL */ +#endif + case 0x0000: /* a dummy value when no SOF driver enabled */ + return 1; + } + } + return 0; +} + /* * initialize the PCI registers */ @@ -1002,6 +1035,10 @@ static int skl_probe(struct pci_dev *pci, dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); return -ENODEV; } + if (skl_sof_support(pci)) { + dev_info(&pci->dev, "SOF driver is preferred on this platform, aborting probe\n"); + return -ENODEV; + } dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: -- 2.20.1

4 10

[alsa-devel] [PATCH] ASoC: amd: acp3x: clean up an indentation issue
by Colin King 27 Sep '19

27 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> There is a return statement that is indented too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index bc4dfafdfcd1..ea57088d50ce 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -631,7 +631,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); - return -ENODEV; + return -ENODEV; } adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); -- 2.20.1

3 2

[alsa-devel] sof-skl_hda: remove volume setting in enable/disable sequences
by GitHub pull_request - opened 27 Sep '19

27 Sep '19

alsa-project/alsa-ucm-conf pull request #2 was opened from juimonen: If the volume setting is defined in enable sequence you will always get that exact volume when switching outputs in some user space software like Pulseaudio. Pulseaudio will itself keep track of the volume previously set. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/2 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/2.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] Playback volumes
by GitHub pull_request - opened 27 Sep '19

27 Sep '19

alsa-project/alsa-ucm-conf pull request #1 was opened from juimonen: Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/1 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [PATCH] ASoC: bdw-rt5677: channel constraint support
by Brent Lu 27 Sep '19

27 Sep '19

BDW boards using this machine driver supports only stereo capture and playback. Implement a constraint to enforce it. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 4a4d335..a312b55 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -22,6 +22,8 @@ #include "../../codecs/rt5677.h" +#define DUAL_CHANNEL 2 + struct bdw_rt5677_priv { struct gpio_desc *gpio_hp_en; struct snd_soc_component *component; @@ -245,6 +247,36 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int bdw_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return 0; +} + +static const struct snd_soc_ops bdw_rt5677_fe_ops = { + .startup = bdw_fe_startup, +}; + /* broadwell digital audio interface glue - connects codec <--> CPU */ SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); @@ -273,6 +305,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { }, .dpcm_capture = 1, .dpcm_playback = 1, + .ops = &bdw_rt5677_fe_ops, SND_SOC_DAILINK_REG(fe, dummy, platform), }, -- 2.7.4

4 9

[alsa-devel] [PATCH 0/4] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 27 Sep '19

27 Sep '19

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the following patch: 1. this series add support reset-controller in infra [v5,2/2] clk: reset: Modify reset-controller driver *** BLURB HERE *** Jiaxin Yu (2): WIP: dt-bindings: medaitek: mt8183: add property "resets" && "reset-names" WIP: ASoC: mt8183: fix audio playback slowly after playback during bootup yong.liang (2): WIP: dt-bindings: mediatek: mt8183: Add #reset-cells WIP: watchdog: mtk_wdt: mt8183: Add reset controller .../bindings/sound/mt8183-afe-pcm.txt | 6 + .../devicetree/bindings/watchdog/mtk-wdt.txt | 9 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 110 +++++++++++++++++- .../reset-controller/mt8183-resets.h | 13 +++ sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 15 +++ 6 files changed, 150 insertions(+), 4 deletions(-) -- 2.18.0

1 4

[alsa-devel] [PATCH] ALSA: hda - Force polling mode on CML for fixing codec communication
by Sathyanarayana Nujella 26 Sep '19

26 Sep '19

We observed the same issue as reported by commit a8d7bde23e7130686b7662 ("ALSA: hda - Force polling mode on CFL for fixing codec communication") Apply the same fix to CML. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Joyce Toh <joyce.toh(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 91e71be42fa4..e6997a5bf098 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -370,6 +370,7 @@ enum { #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) #define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8) +#define IS_CML(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x02c8) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -1754,7 +1755,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_UC; /* Workaround for a communication error on CFL (bko#199007) and CNL */ - if (IS_CFL(pci) || IS_CNL(pci)) + if (IS_CFL(pci) || IS_CNL(pci) || IS_CML(pci)) azx_bus(chip)->polling_mode = 1; if (chip->driver_type == AZX_DRIVER_NVIDIA) { -- 2.7.4

4 3

[alsa-devel] [PATCH] ASoC: topology: Fix a signedness bug in soc_tplg_dapm_widget_create()
by Dan Carpenter 26 Sep '19

26 Sep '19

The "template.id" variable is an enum and in this context GCC will treat it as an unsigned int so it can never be less than zero. Fixes: 8a9782346dcc ("ASoC: topology: Add topology core") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index aa9a1fca46fa..0fd032914a31 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1582,7 +1582,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, /* map user to kernel widget ID */ template.id = get_widget_id(le32_to_cpu(w->id)); - if (template.id < 0) + if ((int)template.id < 0) return template.id; /* strings are allocated here, but used and freed by the widget */ -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: amd: Missing Initialization of IRQFLAGS
by Ravulapati Vishnu vardhan rao 26 Sep '19

26 Sep '19

Fix for missing initialization of IRQFLAGS in ACP-PCI driver and Missing Macro of ACP3x_DEVS. Follow up to IDb33df346 Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 627798a..3887ea0 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -219,7 +219,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, sizeof(struct resource) * 4, GFP_KERNEL); adata->cell = devm_kzalloc(&pci->dev, - sizeof(struct mfd_cell) * 3, + sizeof(struct mfd_cell) * ACP3x_DEVS, GFP_KERNEL); if (!adata->cell) { ret = -ENOMEM; @@ -260,6 +260,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, adata->res[3].flags = IORESOURCE_IRQ; adata->res[3].start = pci->irq; adata->res[3].end = adata->res[3].start; + irqflags = 0; adata->acp3x_audio_mode = ACP3x_I2S_MODE; @@ -282,8 +283,9 @@ static int snd_acp3x_probe(struct pci_dev *pci, adata->cell[2].platform_data = &i2s_pdata[1]; adata->cell[2].pdata_size = sizeof(struct i2s_platform_data); - r = mfd_add_hotplug_devices(adata->parent, adata->cell, 3); - for (i = 0; i < 3 ; i++) + r = mfd_add_hotplug_devices(adata->parent, adata->cell, + ACP3x_DEVS); + for (i = 0; i < ACP3x_DEVS ; i++) dev = get_mfd_cell_dev(adata->cell[i].name, i); break; -- 2.7.4

2 1

[alsa-devel] [PATCH] dt-bindings: sound: Convert Samsung Exynos Odroid XU3/XU4 audio complex to dt-schema
by Marek Szyprowski 26 Sep '19

26 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec to newer dt-schema format. 'clocks' property is unneeded in the bindings and is left undefined in 'properties'. 'samsung,audio-widgets' and 'samsung,audio-routing' are optional from driver perspective and they are set as unrequired. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,odroid.txt | 54 ----------- .../bindings/sound/samsung,odroid.yaml | 91 +++++++++++++++++++ 2 files changed, 91 insertions(+), 54 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt deleted file mode 100644 index e9da2200e173..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ /dev/null @@ -1,54 +0,0 @@ -Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec - -Required properties: - - - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, - "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated), - "samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated), - "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) - - model - the user-visible name of this sound complex - - clocks - should contain entries matching clock names in the clock-names - property - - samsung,audio-widgets - this property specifies off-codec audio elements - like headphones or speakers, for details see widgets.txt - - 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 MAX98090's pins (as - documented in its binding), and the jacks on the board - - For Odroid X2: - "Headphone Jack", "Mic Jack", "DMIC" - - For Odroid U3, XU3: - "Headphone Jack", "Speakers" - - For Odroid XU4: - no entries - -Required sub-nodes: - - - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S - controller - - 'codec' subnode with a 'sound-dai' property containing list of phandles - to the CODEC nodes, first entry must be corresponding to the MAX98090 - CODEC and the second entry must be the phandle of the HDMI IP block node - -Example: - -sound { - compatible = "hardkernel,odroid-xu3-audio"; - model = "Odroid-XU3"; - samsung,audio-routing = - "Headphone Jack", "HPL", - "Headphone Jack", "HPR", - "IN1", "Mic Jack", - "Mic Jack", "MICBIAS"; - - cpu { - sound-dai = <&i2s0 0>; - }; - codec { - sound-dai = <&hdmi>, <&max98090>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml new file mode 100644 index 000000000000..db6d3ea3180e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,odroid.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + oneOf: + - const: hardkernel,odroid-xu3-audio + + - const: hardkernel,odroid-xu4-audio + deprecated: true + + - const: samsung,odroid-xu3-audio + deprecated: true + + - const: samsung,odroid-xu4-audio + deprecated: true + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandles to the I2S controllers + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles to the CODEC nodes, + first entry must be corresponding to the MAX98090 CODEC and + the second entry must be the phandle of the HDMI IP block node. + + samsung,audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + This property specifies off-codec audio elements + like headphones or speakers, for details see widgets.txt + + 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 MAX98090's pins (as + documented in its binding), and the jacks on the board. + For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC" + For Odroid U3, XU3: "Headphone Jack", "Speakers" + For Odroid XU4: no entries + +required: + - compatible + - model + - cpu + - codec + +examples: + - | + sound { + compatible = "hardkernel,odroid-xu3-audio"; + model = "Odroid-XU3"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&hdmi>, <&max98090>; + }; + }; + -- 2.17.1

2 1

[alsa-devel] [PATCH] dt-bindings: sound: Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to dt-schema
by Marek Szyprowski 26 Sep '19

26 Sep '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> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,tm2-audio.txt | 42 ---------- .../bindings/sound/samsung,tm2-audio.yaml | 83 +++++++++++++++++++ 2 files changed, 83 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..377f8cbe17b8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml @@ -0,0 +1,83 @@ +# 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: + maxItems: 1 + + 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. + + i2s-controller: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: phandle of the I2S0. + - description: phandle of the I2S1. + + audio-amplifier: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the MAX98504 amplifier. + + 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 and MAX98504's pins and the jacks on the + board: HP, SPK, Main Mic, Sub Mic, Third Mic, Headset Mic. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + maxItems: 1 + +required: + - compatible + - model + - audio-codec + - i2s-controller + - audio-amplifier + - samsung,audio-routing + - mic-bias-gpios + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,tm2-audio"; + model = "wm5110"; + audio-codec = <&wm5110>, <&hdmi>; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + 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] [PATCH V2 1/2] string: Add stracpy and stracpy_pad mechanisms
by Rasmus Villemoes 26 Sep '19

26 Sep '19

On 26/09/2019 10.25, Stephen Kitt wrote: > Le 26/09/2019 09:29, Rasmus Villemoes a écrit : >> On 26/09/2019 02.01, Stephen Kitt wrote: >>> Le 25/09/2019 23:50, Andrew Morton a écrit : >>>> On Tue, 23 Jul 2019 06:51:36 -0700 Joe Perches <joe(a)perches.com> wrote: >>>> >> >> Please don't. At least not for the cases where the source is a string >> literal - that just gives worse code generation (because gcc doesn't >> know anything about strscpy or strlcpy), and while a run-time (silent) >> truncation is better than a run-time buffer overflow, wouldn't it be >> even better with a build time error? > > Yes, that was the plan once Joe's patch gets merged (if it does), and my > patch was only an example of using stracpy, as a step on the road. I was > intending to follow up with a patch converting stracpy to something like > https://www.openwall.com/lists/kernel-hardening/2019/07/06/14 > > __FORTIFY_INLINE ssize_t strscpy(char *dest, const char *src, size_t count) > { > size_t dest_size = __builtin_object_size(dest, 0); > size_t src_size = __builtin_object_size(src, 0); > if (__builtin_constant_p(count) && > __builtin_constant_p(src_size) && > __builtin_constant_p(dest_size) && Eh? Isn't the output of __builtin_object_size() by definition a compile-time constant - whatever the compiler happens to know about the object size (or a sentinel 0 or -1 depending on the type argument)? > > #define stracpy(dest, src) \ > ({ \ > size_t count = ARRAY_SIZE(dest); \ > size_t dest_size = __builtin_object_size(dest, 0); \ > size_t src_size = __builtin_object_size(src, 0); \ > BUILD_BUG_ON(!(__same_type(dest, char[]) || \ > __same_type(dest, unsigned char[]) || \ > __same_type(dest, signed char[]))); \ > \ > (__builtin_constant_p(count) && \ > __builtin_constant_p(src_size) && \ > __builtin_constant_p(dest_size) && \ > src_size <= count && \ > src_size <= dest_size && \ > src[src_size - 1] == '\0') ? \ > (((size_t) strcpy(dest, src)) & 0) + src_size - 1 \ > : \ > strscpy(dest, src, count); \ > }) > > and both of these get optimised to movs when copying a constant string > which fits in the target. But does it catch the case of overflowing a char[] member in a struct passed by reference at build time? I'm surprised that __builtin_object_size(dest, 0) seems to be (size_t)-1, when dest is s->name (with struct s { char name[4]; };). So I'm not very confident that any of the fancy fortify logic actually works here - we _really_ should have some Kbuild infrastructure for saying "this .c file should not compile" so we can test that the fortifications actually work in the simple and common cases. > I was going at this from the angle of improving the existing APIs and > their resulting code. I'm not against stracpy() as a wrapper for strscpy(), but we should make sure that whenever we can fail at build time (i.e., both source and dst lengths known), we do - and in that case also let the compiler optimize the copy (not only to do the immediate movs, but that also gives it wider opportunity to remove it completely as dead stores if the surrounding code ends up dead - with a call to some strscpy(), gcc cannot eliminate that). If stracpy() can be made sufficiently magic that it fails at build time for the string literal cases, fine, let's not add yet another API. Otherwise, I think the static_strcpy() is a much more readable and reliable API for those cases. If I'm reading your stracpy() macro correctly, you're explicitly requesting a run-time truncation (the src_size <= dest_size check causing as to fall back to strscpy) rather than failing at build time. Rasmus

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Re: [alsa-devel] [PATCH V2 1/2] string: Add stracpy and stracpy_pad mechanisms
by Rasmus Villemoes 26 Sep '19

26 Sep '19

On 26/09/2019 02.01, Stephen Kitt wrote: > Le 25/09/2019 23:50, Andrew Morton a écrit : >> On Tue, 23 Jul 2019 06:51:36 -0700 Joe Perches <joe(a)perches.com> wrote: >> >>> Several uses of strlcpy and strscpy have had defects because the >>> last argument of each function is misused or typoed. >>> >>> Add macro mechanisms to avoid this defect. >>> >>> stracpy (copy a string to a string array) must have a string >>> array as the first argument (dest) and uses sizeof(dest) as the >>> count of bytes to copy. >>> >>> These mechanisms verify that the dest argument is an array of >>> char or other compatible types like u8 or s8 or equivalent. >>> >>> A BUILD_BUG is emitted when the type of dest is not compatible. >>> >> >> I'm still reluctant to merge this because we don't have code in -next >> which *uses* it. You did have a patch for that against v1, I believe? >> Please dust it off and send it along? > > Joe had a Coccinelle script to mass-convert strlcpy and strscpy. > Here's a different patch which converts some of ALSA's strcpy calls to > stracpy: Please don't. At least not for the cases where the source is a string literal - that just gives worse code generation (because gcc doesn't know anything about strscpy or strlcpy), and while a run-time (silent) truncation is better than a run-time buffer overflow, wouldn't it be even better with a build time error? Something like /** static_strcpy - run-time version of static initialization * * @d: destination array, must be array of char of known size * @s: source, must be a string literal * * This is a simple wrapper for strcpy(d, s), which checks at build-time that the strcpy() won't overflow. In most cases (for short strings), gcc won't even emit a call to strcpy but rather just do a few immediate stores into the destination, e.g. 0: c7 07 66 6f 6f 00 movl $0x6f6f66,(%rdi) * for strcpy(d->name, "foo"). */ #define static_strcpy(d, s) ({ \ static_assert(__same_type(d, char[]), "destination must be char array"); \ static_assert(__same_type(s, const char[], "source must be a string literal"); \ static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ strcpy(d, s); \ }) The "" s "" trick guarantees that s is a string literal - it probably doesn't give a very nice error message, but that's why I added the redundant type comparison to a const char[] which should hopefully give a better clue. Rasmus PS: Yes, we already have a fortified strcpy(), but for some reason it doesn't catch the common case where we're populating a char[] member of some struct. So this diff --git a/lib/vsprintf.c b/lib/vsprintf.c index e78017a3e1bd..3b0c5ae9f49e 100644 --- a/lib/vsprintf.c +++ b/lib/vsprintf.c @@ -3420,3 +3420,14 @@ int sscanf(const char *buf, const char *fmt, ...) return i; } EXPORT_SYMBOL(sscanf); + +struct s { char name[4]; }; +char buf[4]; +void f3(void) +{ + strcpy(buf, "long"); +} +void f4(struct s *s) +{ + strcpy(s->name, "long"); +} with CONFIG_FORTIFY_SOURCE=y complains about f3(), but f4() is just fine... PPS: A quick test of static_strcpy(): // ss.c #include <errno.h> #include <string.h> #include <assert.h> #define __same_type(x, y) __builtin_types_compatible_p(typeof(x), typeof(y)) #define static_strcpy(d, s) ({ \ static_assert(__same_type(d, char[]), "destination must be char array"); \ static_assert(__same_type(s, const char[]), "source must be a string literal"); \ static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ strcpy(d, s); \ }) struct s { char name[4]; }; void f0(struct s *s) { static_strcpy(s->name, "foo"); } #if 1 void f1(struct s *s) { static_strcpy(s->name, strerror(EIO)); } void f2(struct s *s) { static_strcpy(s->name, "long"); } void f3(char *d) { static_strcpy(d, "foo"); } void f4(char *d) { static_strcpy(d, strerror(EIO)); } #endif // gcc -Wall -O2 -o ss.o -c ss.c In file included from ss.c:3:0: ss.c: In function ‘f1’: ss.c:9:3: error: static assertion failed: "source must be a string literal" static_assert(__same_type(s, const char[]), "source must be a string literal"); \ ^ ss.c:18:24: note: in expansion of macro ‘static_strcpy’ void f1(struct s *s) { static_strcpy(s->name, strerror(EIO)); } ^~~~~~~~~~~~~ ss.c:18:47: error: expected ‘)’ before ‘strerror’ void f1(struct s *s) { static_strcpy(s->name, strerror(EIO)); } ^ ss.c:10:40: note: in definition of macro ‘static_strcpy’ static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ ^ In file included from ss.c:3:0: ss.c: In function ‘f2’: ss.c:10:3: error: static assertion failed: "source does not fit in destination" static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ ^ ss.c:19:24: note: in expansion of macro ‘static_strcpy’ void f2(struct s *s) { static_strcpy(s->name, "long"); } ^~~~~~~~~~~~~ ss.c: In function ‘f3’: ss.c:8:3: error: static assertion failed: "destination must be char array" static_assert(__same_type(d, char[]), "destination must be char array"); \ ^ ss.c:20:20: note: in expansion of macro ‘static_strcpy’ void f3(char *d) { static_strcpy(d, "foo"); } ^~~~~~~~~~~~~ ss.c: In function ‘f4’: ss.c:8:3: error: static assertion failed: "destination must be char array" static_assert(__same_type(d, char[]), "destination must be char array"); \ ^ ss.c:21:20: note: in expansion of macro ‘static_strcpy’ void f4(char *d) { static_strcpy(d, strerror(EIO)); } ^~~~~~~~~~~~~ ss.c:9:3: error: static assertion failed: "source must be a string literal" static_assert(__same_type(s, const char[]), "source must be a string literal"); \ ^ ss.c:21:20: note: in expansion of macro ‘static_strcpy’ void f4(char *d) { static_strcpy(d, strerror(EIO)); } ^~~~~~~~~~~~~ ss.c:21:37: error: expected ‘)’ before ‘strerror’ void f4(char *d) { static_strcpy(d, strerror(EIO)); } ^ ss.c:10:40: note: in definition of macro ‘static_strcpy’ static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ ^

1 0

Re: [alsa-devel] [PATCH V5 4/4] ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8
by S.j. Wang 26 Sep '19

26 Sep '19

Hi > Just a small concern... > > On Thu, Sep 26, 2019 at 09:29:51AM +0800, Shengjiu Wang wrote: > > static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) > > { > > + > > + release_pair = false; > > + ret = snd_soc_set_runtime_hwparams(substream, > > + &snd_imx_hardware); > > This set_runtime_hwparams() always returns 0 for now, but if one day it > changes and it fails here, kfree() will be still ignored although the startup() > gets error-out. > > We could avoid this if we continue to ignore the return value like the > current code. Or we may check ret at kfree() also? I like to ignore the return value. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH V5 0/4] update supported sample format
by Shengjiu Wang 26 Sep '19

26 Sep '19

This patch serial is to update the supported format for fsl_asrc and fix some format issue. Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch changes in v3 - Fix build report by kbuild test robot <lkp(a)intel.com> - change snd_dmaengine_pcm_set_runtime_hwparams to snd_dmaengine_pcm_refine_runtime_hwparams changes in v4 - update according to Nicolin's comments. changes in v5 - free asrc pair when error happens include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 66 ++++++++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 61 ++------------------ 6 files changed, 200 insertions(+), 87 deletions(-) -- 2.21.0

2 5

Re: [alsa-devel] [PATCH v2] ASoC: Intel: Skylake: prevent memory leak in snd_skl_parse_uuids
by Pierre-Louis Bossart 25 Sep '19

25 Sep '19

On 9/25/19 11:19 AM, Navid Emamdoost wrote: > In snd_skl_parse_uuids if allocation for module->instance_id fails, the > allocated memory for module shoulde be released. I changes the > allocation for module to use devm_kzalloc to be resource_managed > allocation and avoid the release in error path. if you use devm_, don't you need to fix the error path as well then, I see a kfree(uuid) in skl_freeup_uuid_list(). I am not very familiar with this code but the error seems to be that the list_add_tail() is called after the module->instance_id is allocated, so there is a risk that the module allocated earlier is not freed (since it's not yet added to the list). Freeing the module as done in patch 1 works, using devm_ without fixing the error path does not seem correct to me. > > Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> > --- > Changes in v2: > - Changed the allocation for module from kzalloc to devm_kzalloc > --- > sound/soc/intel/skylake/skl-sst-utils.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c > index d43cbf4a71ef..ac37f04b0eea 100644 > --- a/sound/soc/intel/skylake/skl-sst-utils.c > +++ b/sound/soc/intel/skylake/skl-sst-utils.c > @@ -284,7 +284,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, > */ > > for (i = 0; i < num_entry; i++, mod_entry++) { > - module = kzalloc(sizeof(*module), GFP_KERNEL); > + module = devm_kzalloc(ctx->dev, sizeof(*module), GFP_KERNEL); > if (!module) { > ret = -ENOMEM; > goto free_uuid_list; >

1 0

[alsa-devel] [PATCH] [MT6660] Mediatek Smart Amplifier Driver
by richtek.jeff.chang＠gmail.com 25 Sep '19

25 Sep '19

From: Richtek Jeff Chang <richtek.jeff.chang(a)gmail.com> Signed-off-by: Richtek Jeff Chang <richtek.jeff.chang(a)gmail.com> --- sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 802 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6660.h | 68 ++++ 4 files changed, 885 insertions(+) create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f89a53..3407436 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -120,6 +120,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1444,6 +1445,18 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "MediaTek MT6660 Speaker amplifier" + depends on I2C + select CRC32 + select CRYPTO_SHA256 + select CRYPTO_RSA + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b4bb8c..e6a8054 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -117,6 +117,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -398,6 +399,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..b82cf03 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,802 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +union mt6660_multi_byte_data { + u32 data_u32; + u16 data_u16; + u8 data_u8; + u8 data[4]; +}; + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2}, + { MT6660_REG_TDM_CFG3, 2}, + { MT6660_REG_HCLIP_CTRL, 2}, + { MT6660_REG_DA_GAIN, 2}, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +static int32_t mt6660_i2c_update_bits(struct mt6660_chip *chip, + uint32_t addr, uint32_t mask, uint32_t data) +{ + int ret = 0; + uint32_t value; + int size = mt6660_get_reg_size(addr); + union mt6660_multi_byte_data mdata; + + memcpy(&mdata, &data, sizeof(uint32_t)); + + mutex_lock(&chip->io_lock); + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, addr, size, (u8 *)&value); + if (ret < 0) { + mutex_unlock(&chip->io_lock); + return ret; + } + switch (size) { + case 1: + value &= ~mask; + value |= (mdata.data_u8 & mask); + break; + case 2: + value = be16_to_cpu(value); + value &= ~mask; + value |= (mdata.data_u16 & mask); + value = be16_to_cpu(value); + break; + case 4: + value = be32_to_cpu(value); + value &= ~mask; + value |= (mdata.data_u32 & mask); + value = be32_to_cpu(value); + break; + default: + dev_err(chip->dev, "%s Invalid bytes\n", __func__); + break; + } + + ret = i2c_smbus_write_i2c_block_data( + chip->i2c, addr, size, (u8 *)&value); + if (ret < 0) { + mutex_unlock(&chip->io_lock); + return ret; + } + mutex_unlock(&chip->io_lock); + + return 0; +} + +static int mt6660_i2c_read(struct mt6660_chip *chip, unsigned int reg) +{ + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + return reg_data; +} + +static unsigned int mt6660_component_io_read( + struct snd_soc_component *component, unsigned int reg) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + return mt6660_i2c_read(chip, reg); +} + +static int mt6660_component_io_write(struct snd_soc_component *component, + unsigned int reg, unsigned int data) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (data >> (8 * i)) & 0xff; + + return i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); +} + +static const int mt6660_dump_table[] = { + MT6660_REG_DEVID, + MT6660_REG_SYSTEM_CTRL, + MT6660_REG_IRQ_STATUS1, + MT6660_REG_SERIAL_CFG1, + MT6660_REG_DATAO_SEL, + MT6660_REG_TDM_CFG3, + MT6660_REG_HPF_CTRL, + MT6660_REG_HPF1_COEF, + MT6660_REG_HPF2_COEF, + MT6660_REG_PATH_BYPASS, + MT6660_REG_WDT_CTRL, + MT6660_REG_HCLIP_CTRL, + MT6660_REG_VOL_CTRL, + MT6660_REG_SPS_CTRL, + MT6660_REG_SIGMAX, + MT6660_REG_CALI_T0, + MT6660_REG_BST_CTRL, + MT6660_REG_PROTECTION_CFG, + MT6660_REG_DA_GAIN, + MT6660_REG_AUDIO_IN2_SEL, + MT6660_REG_SIG_GAIN, + MT6660_REG_PLL_CFG1, + MT6660_REG_DRE_CTRL, + MT6660_REG_DRE_THDMODE, + MT6660_REG_DRE_CORASE, + MT6660_REG_PWM_CTRL, + MT6660_REG_DC_PROTECT_CTRL, + MT6660_REG_ADC_USB_MODE, + MT6660_REG_INTERNAL_CFG, + MT6660_REG_RESV0, + MT6660_REG_RESV1, + MT6660_REG_RESV2, + MT6660_REG_RESV3, + MT6660_REG_RESV7, + MT6660_REG_RESV10, + MT6660_REG_RESV11, + MT6660_REG_RESV16, + MT6660_REG_RESV17, + MT6660_REG_RESV19, + MT6660_REG_RESV21, + MT6660_REG_RESV23, + MT6660_REG_RESV31, + MT6660_REG_RESV40, +}; + +static const struct codec_reg_val e4_reg_inits[] = { + { MT6660_REG_WDT_CTRL, 0x80, 0x00 }, + { MT6660_REG_SPS_CTRL, 0x01, 0x01 }, + { MT6660_REG_AUDIO_IN2_SEL, 0x1c, 0x04 }, + { MT6660_REG_RESV11, 0x0c, 0x00 }, + { MT6660_REG_RESV31, 0x03, 0x03 }, + { MT6660_REG_RESV40, 0x01, 0x00 }, + { MT6660_REG_RESV0, 0x44, 0x04 }, + { MT6660_REG_RESV19, 0xff, 0x82 }, + { MT6660_REG_RESV17, 0x7777, 0x7273 }, + { MT6660_REG_RESV16, 0x07, 0x03 }, + { MT6660_REG_DRE_CORASE, 0xe0, 0x20 }, + { MT6660_REG_ADDA_CLOCK, 0xff, 0x70 }, + { MT6660_REG_RESV21, 0xff, 0x20 }, + { MT6660_REG_DRE_THDMODE, 0xff, 0x40 }, + { MT6660_REG_RESV23, 0xffff, 0x17f8 }, + { MT6660_REG_PWM_CTRL, 0xff, 0x15 }, + { MT6660_REG_ADC_USB_MODE, 0xff, 0x00 }, + { MT6660_REG_PROTECTION_CFG, 0xff, 0x1d }, + { MT6660_REG_HPF1_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_HPF2_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_SIG_GAIN, 0xff, 0x58 }, + { MT6660_REG_RESV6, 0xff, 0xce }, + { MT6660_REG_SIGMAX, 0xffff, 0x7fff }, + { MT6660_REG_DA_GAIN, 0xffff, 0x0116 }, + { MT6660_REG_TDM_CFG3, 0x1800, 0x0800 }, + { MT6660_REG_DRE_CTRL, 0x1f, 0x07 }, +}; + +static int mt6660_i2c_init_setting(struct mt6660_chip *chip) +{ + int i, len, ret; + const struct codec_reg_val *init_table; + + init_table = e4_reg_inits; + len = ARRAY_SIZE(e4_reg_inits); + + for (i = 0; i < len; i++) { + ret = mt6660_i2c_update_bits(chip, init_table[i].addr, + init_table[i].mask, init_table[i].data); + if (ret < 0) + return ret; + } + return 0; +} + +static int mt6660_chip_power_on( + struct snd_soc_component *component, int on_off) +{ + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = 0; + unsigned int val; + + dev_dbg(component->dev, "%s: on_off = %d\n", __func__, on_off); + mutex_lock(&chip->var_lock); + if (on_off) { + if (chip->pwr_cnt == 0) { + ret = mt6660_i2c_update_bits(chip, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); + val = mt6660_i2c_read(chip, MT6660_REG_IRQ_STATUS1); + dev_info(chip->dev, + "%s reg0x05 = 0x%x\n", __func__, val); + } + chip->pwr_cnt++; + } else { + chip->pwr_cnt--; + if (chip->pwr_cnt == 0) { + ret = mt6660_i2c_update_bits(chip, + MT6660_REG_SYSTEM_CTRL, 0x01, 0xff); + } + if (chip->pwr_cnt < 0) { + dev_warn(chip->dev, "not paired on/off\n"); + chip->pwr_cnt = 0; + } + } + mutex_unlock(&chip->var_lock); + if (ret < 0) + pr_err("%s ret = %d\n", __func__, ret); + return ret; +} + +static int mt6660_component_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret; + unsigned int val; + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + if (dapm->bias_level == level) { + dev_warn(component->dev, "%s: repeat level change\n", __func__); + return 0; + } + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level != SND_SOC_BIAS_OFF) + break; + dev_dbg(component->dev, "exit low power mode\n"); + ret = mt6660_chip_power_on(component, 1); + if (ret < 0) { + dev_err(component->dev, "power on fail\n"); + return ret; + } + break; + case SND_SOC_BIAS_OFF: + dev_dbg(component->dev, "enter low power mode\n"); + val = mt6660_i2c_read(chip, MT6660_REG_IRQ_STATUS1); + dev_info(component->dev, + "%s reg0x05 = 0x%x\n", __func__, val); + ret = mt6660_chip_power_on(component, 0); + if (ret < 0) { + dev_err(component->dev, "power off fail\n"); + return ret; + } + break; + default: + return -EINVAL; + } + dapm->bias_level = level; + dev_dbg(component->dev, "c bias_level = %d\n", level); + return 0; +} + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + ret = mt6660_component_set_bias_level(component, SND_SOC_BIAS_STANDBY); + if (ret < 0) { + dev_err(component->dev, "config bias standby fail\n"); + return ret; + } + + ret = mt6660_component_set_bias_level(component, SND_SOC_BIAS_OFF); + if (ret < 0) { + dev_err(component->dev, "config bias off fail\n"); + return ret; + } + chip->component = component; + return 0; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "%s++\n", __func__); + chip->component = NULL; + dev_dbg(component->dev, "%s--\n", __func__); +} + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_PRE_PMD: + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SPK("SPK", NULL), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback"}, + { "PGA", NULL, "DAC"}, + { "ClassD", NULL, "PGA"}, + { "SPK", NULL, "ClassD"}, + { "VI ADC", NULL, "ClassD"}, + { "aif_capture", NULL, "VI ADC"}, +}; + +static int mt6660_component_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + int ret, put_ret = 0; + + ret = mt6660_chip_power_on(component, 1); + if (ret < 0) + dev_err(component->dev, "%s: pwr on fail\n", __func__); + put_ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + ret = mt6660_chip_power_on(component, 0); + if (ret < 0) + dev_err(component->dev, "%s: pwr off fail\n", __func__); + return put_ret; +} + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip_Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Volume_Ctrl", MT6660_REG_VOL_CTRL, 0, 255, + 1, snd_soc_get_volsw, mt6660_component_put_volsw, + vol_ctl_tlv), + SOC_SINGLE_EXT("WDT_Enable", MT6660_REG_WDT_CTRL, 7, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Hard_Clip_Enable", MT6660_REG_HCLIP_CTRL, 8, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Clip_Enable", MT6660_REG_SPS_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("BoostMode", MT6660_REG_BST_CTRL, 0, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DRE_Enable", MT6660_REG_DRE_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DC_Protect_Enable", + MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("I2SLRS", MT6660_REG_DATAO_SEL, 6, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("I2SDOLS", MT6660_REG_DATAO_SEL, 3, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("I2SDORS", MT6660_REG_DATAO_SEL, 0, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + /* for debug purpose */ + SOC_SINGLE_EXT("HPF_AUD_IN_EN", MT6660_REG_HPF_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("AUD_LOOP_BACK", MT6660_REG_PATH_BYPASS, 4, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Mute_Enable", MT6660_REG_SYSTEM_CTRL, 1, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("CS_Comp_Disable", MT6660_REG_PATH_BYPASS, 2, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("T0_SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip_Rev", SND_SOC_NOPM, 0, 16, 0, + mt6660_component_get_volsw, NULL), +}; + + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + + .read = mt6660_component_io_read, + .write = mt6660_component_io_write, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .set_bias_level = mt6660_component_set_bias_level, + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(dai->component); + int ret = 0; + + dev_dbg(dai->dev, "%s\n", __func__); + if (dapm->bias_level == SND_SOC_BIAS_OFF) + ret = mt6660_component_set_bias_level(dai->component, + SND_SOC_BIAS_STANDBY); + return ret; +} + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .startup = mt6660_component_aif_startup, + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int mt6660_chip_id_check(struct i2c_client *i2c) +{ + u8 id[2] = {0}; + int ret = 0; + + i2c_smbus_write_byte_data(i2c, 0x03, 0x00); + ret = i2c_smbus_read_i2c_block_data(i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) + return -ENODEV; + i2c_smbus_write_byte_data(i2c, 0x03, 0x01); + return 0; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + i2c_smbus_write_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return i2c_smbus_write_byte_data( + chip->i2c, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +int mt6660_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + ret = mt6660_chip_id_check(client); + if (ret < 0) { + dev_err(&client->dev, "chip id check fail\n"); + return ret; + } + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->var_lock); + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 1 fail\n"); + goto probe_fail; + } + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + ret = mt6660_i2c_init_setting(chip); + if (ret < 0) { + dev_err(chip->dev, "chip i2c init setting fail\n"); + goto probe_fail; + } + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(chip->dev, "chip power off fail\n"); + goto probe_fail; + } + return snd_soc_register_component(chip->dev, + &mt6660_component_driver, &mt6660_codec_dai, 1); +probe_fail: + mutex_destroy(&chip->var_lock); + return ret; +} + +int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + dev_dbg(chip->dev, "%s++\n", __func__); + snd_soc_unregister_component(chip->dev); + mutex_destroy(&chip->var_lock); + dev_dbg(chip->dev, "%s--\n", __func__); + return 0; +} + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(mt6660_of_id), + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.6_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..4d5cc16 --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,68 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct snd_soc_component *component; + struct platform_device *param_dev; + struct mutex var_lock; + struct mutex io_lock; + u16 chip_rev; + int pwr_cnt; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

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[alsa-devel] About soc_check_tplg_fes
by Daniel Baluta 25 Sep '19

25 Sep '19

Hi all, In function soc_check_tplg_fes (sound/soc/soc-core.c) there is this code: for_each_card_prelinks(card, i, dai_link) { /* ignore this FE */ if (dai_link->dynamic) { dai_link->ignore = true; continue; } dev_info(card->dev, "info: override FE DAI link %s\n", card->dai_link[i].name); So, we ignore any FE DAI link but then the info message says the contrary. Any idea? thanks, Daniel.

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[alsa-devel] [PATCH v3 0/2] Add CML M/C for RT5682 and RT1011
by sathya.prakash.m.r＠intel.com 25 Sep '19

25 Sep '19

From: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Naveen Manohar (2): ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 ASoC: Intel: Add acpi match for rt1011 based m/c driver sound/soc/intel/boards/Kconfig | 18 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 473 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 500 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- Changes from V1: Addressed comments on ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 Changes from V2: Reordered acpi match table as MAX98357A is a non I2C device. Hence probe order might fail. 1.9.1

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[alsa-devel] [RFC] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 25 Sep '19

25 Sep '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Start DMA first followed by CPU DAI (currently used sequence) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Stop CPU DAI first followed by DMA Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, on a new TI platform (j721e) where we can disable the McASP FIFO (the CPU dai) since the UDMA + PDMA provides the buffering I have started to see error interrupts right after pcm_trigger:STOP and in rare cases even on PAUSE that McASP underruns. I was also able to reproduce the same issue on am335x board, but it is much harder to trigger it. With this patch the underrun after trigger:STOP is gone. If I think about the issue, I'm not sure why it was not noticed before as the behavior makes sense: we stop the DMA first then we stop the CPU DAI. If between the DMA stop and DAI stop we would need a sample in the DAI (which is still running) then for sure we will underrun in the HW (or overrun in case of capture). When I run the ALSA conformance test [1] it is easier to trigger. Not sure if anyone else have seen such underrun/overrun when stopping a stream, but the fact that I have seen it with both UDMA+PDMA and EDMA on different platforms makes me wonder if the issue can be seen on other platforms as well. [1] https://chromium.googlesource.com/chromiumos/platform/audiotest/+/master/al… Regards, Peter --- sound/soc/soc-pcm.c | 66 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 62 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..c96430e70752 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,24 +1056,60 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_trigger(codec_dai, substream, cmd); if (ret < 0) return ret; } - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; + snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; - ret = snd_soc_component_trigger(component, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 25 Sep '19

25 Sep '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/soc-pcm.c | 66 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 62 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..41e081e95108 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1070,10 +1070,46 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - snd_soc_dai_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- 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] [PATCH] ASoC: Intel: Skylake: prevent memory leak in snd_skl_parse_uuids
by Andy Shevchenko 25 Sep '19

25 Sep '19

On Tue, Sep 24, 2019 at 11:08:38PM -0500, Navid Emamdoost wrote: > In snd_skl_parse_uuids if allocation for module->instance_id fails, the > allocated memory for module shoulde be released. Since it's using devm_*() for the second allocation it would be cleaner to either switch it to regular kcalloc() or to switch the first one to devm_kzalloc(), whatever suits better. > module->instance_id = devm_kzalloc(ctx->dev, size, GFP_KERNEL); > if (!module->instance_id) { > + kfree(module); -- With Best Regards, Andy Shevchenko

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[alsa-devel] [PATCH 0/3] Make pcm_ioplug check lock status before locking (fixes pcm_jack lockups)
by Ben Russell 25 Sep '19

25 Sep '19

This is my first time contributing a patch to a mailing list. If I've made a mess, please let me know so I can learn how to avoid it in future. The purpose of this patchset is to fix a specific common lockup in the pcm_jack plugin (from alsa-plugins). I'm not sure if this is the right approach to take, but at the very least it should make the pcm_ioplug code a bit more resilient. The problem is this: When using the pcm_jack plugin, if a program attempts to play audio using the SND_PCM_FORMAT_FLOAT format, said program locks up. This should be enough to reproduce the bug: pcm.rawjack { type jack playback_ports { 0 system:playback_1 1 system:playback_2 } capture_ports { 0 system:capture_1 1 system:capture_2 } } pcm.!default { type plug slave.pcm "rawjack" } What's happening is that several snd_pcm_ioplug_* functions assume that the pcm mutex is locked already. It then proceeds to unlock the mutex, call a function, and then relock the mutex. When the mutex isn't locked already, the initial unlock results in a silently ignored pthread error, and the lock results in the program eventually deadlocking as it doesn't expect the mutex to be held at that point. Patch 2 modifies pcm_ioplug to check if the mutex is held before doing the unlock-act-lock sequence, and if the mutex is not held then it skips the unlock and lock stages. This depends on Patch 1, which adds a snd_pcm_is_locked function to give the state of the mutex. Patch 3 is completely optional. It adds assertions which make sure that all uses of snd_pcm_lock/snd_pcm_unlock are correct. On one hand this will likely result in crashes in some of the less refined parts of the code. On the other hand, when that happens, you'll know which parts need a bit more love. I know it was useful for finding this issue in the first place. These patches fix the problems I am having, but if you have a more suitable approach to fixing this problem then please let me know. Regards, Ben R Ben Russell (3): pcm_local: Add snd_pcm_is_locked pcm_ioplug: Don't unlock+lock if we're not locked pcm_local: assert() when using mutexes incorrectly src/pcm/pcm_ioplug.c | 50 +++++++++++++++++++++++----------- src/pcm/pcm_local.h | 64 ++++++++++++++++++++++++++++++++++++++------ 2 files changed, 91 insertions(+), 23 deletions(-) -- 2.23.0

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[alsa-devel] [PATCH v2 0/2] Add CML M/C for RT5682 and RT1011
by sathya.prakash.m.r＠intel.com 25 Sep '19

25 Sep '19

From: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Naveen Manohar (2): ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 ASoC: Intel: Add acpi match for rt1011 based m/c driver sound/soc/intel/boards/Kconfig | 18 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 473 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 500 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- Changes from V1: Addressed comments on ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 1.9.1

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Re: [alsa-devel] [PATCH V4 4/4] ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8
by S.j. Wang 25 Sep '19

25 Sep '19

Hi > On Tue, Sep 24, 2019 at 06:52:35PM +0800, Shengjiu Wang wrote: > > There is error "aplay: pcm_write:2023: write error: Input/output error" > > on i.MX8QM/i.MX8QXP platform for S24_3LE format. > > > > In i.MX8QM/i.MX8QXP, the DMA is EDMA, which don't support 24bit > > sample, but we didn't add any constraint, that cause issues. > > > > So we need to query the caps of dma, then update the hw parameters > > according to the caps. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/fsl_asrc.c | 4 +-- > > sound/soc/fsl/fsl_asrc.h | 3 ++ > > sound/soc/fsl/fsl_asrc_dma.c | 59 > > +++++++++++++++++++++++++++++++----- > > 3 files changed, 56 insertions(+), 10 deletions(-) > > > > @@ -270,12 +268,17 @@ static int fsl_asrc_dma_hw_free(struct > > snd_pcm_substream *substream) > > > > static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) > > { > > + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; > > struct snd_soc_pcm_runtime *rtd = substream->private_data; > > struct snd_pcm_runtime *runtime = substream->runtime; > > struct snd_soc_component *component = > snd_soc_rtdcom_lookup(rtd, > > DRV_NAME); > > + struct snd_dmaengine_dai_dma_data *dma_data; > > struct device *dev = component->dev; > > struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); > > struct fsl_asrc_pair *pair; > > + struct dma_chan *tmp_chan = NULL; > > + u8 dir = tx ? OUT : IN; > > + int ret = 0; > > > > pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); > > Sorry, I didn't catch it previously. We would need to release this memory > also for all error-out paths, as the code doesn't have any error-out routine, > prior to applying this change. > > > if (!pair) > > @@ -285,11 +288,51 @@ static int fsl_asrc_dma_startup(struct > > snd_pcm_substream *substream) > > > + /* Request a dummy pair, which will be released later. > > + * Request pair function needs channel num as input, for this > > + * dummy pair, we just request "1" channel temporary. > > + */ > > "temporary" => "temporarily" > > > + ret = fsl_asrc_request_pair(1, pair); > > + if (ret < 0) { > > + dev_err(dev, "failed to request asrc pair\n"); > > + return ret; > > + } > > + > > + /* Request a dummy dma channel, which will be release later. */ > > "release" => "released" Ok, will update them. Best regards Wang shengjiu

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[alsa-devel] [PATCH V4 0/4] update supported sample format
by Shengjiu Wang 25 Sep '19

25 Sep '19

This patch serial is to update the supported format for fsl_asrc and fix some format issue. Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch changes in v3 - Fix build report by kbuild test robot <lkp(a)intel.com> - change snd_dmaengine_pcm_set_runtime_hwparams to snd_dmaengine_pcm_refine_runtime_hwparams changes in v4 - update according to Nicolin's comments. include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 59 ++++++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 61 ++------------------ 6 files changed, 193 insertions(+), 87 deletions(-) -- 2.21.0

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[alsa-devel] Making alsamixer configurable, adding mouse support + minor fixes
by GitHub pull_request - opened 24 Sep '19

24 Sep '19

alsa-project/alsa-utils pull request #26 was opened from braph: Hello, I'm using the alsamixer, but I am missing mouse support and the ability to configure it. I implemented those things for myself now, maybe it would also be useful for others. There are some commits that would even fit in the original alsamixer code without my rather heavy modifications, these are: 6eb0bc8f0848567f302e94ea5ee3a6d7c6966e65: _alsamixer: remove redundant sprintf() calls_ bef5b723a43f310ba73bcaa52f3da74ef5e08dde: _alsamixer: use mvwhline/mvwvline for drawing lines_ ae0604c93aa4bacbb678eb8ec832ad9e17c8b56a: _alsamixer: device_name.c: recognize DEL (ASCII 127) as backspace_ And maybe 485f3b0da1ffead2f88cffd9f2203516000415b5: _alsamixer: fix window_size_changed()_ Regards, Benjamin Request URL : https://github.com/alsa-project/alsa-utils/pull/26 Patch URL : https://github.com/alsa-project/alsa-utils/pull/26.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] alsa-ucm-conf repository - initial import
by Jaroslav Kysela 24 Sep '19

24 Sep '19

Hello all, I just made an initial import to to alsa-ucm-conf repository which is in the 'import' branch: https://github.com/alsa-project/alsa-ucm-conf/tree/import I do not have the BSD-3-Clause licence acknowledgements from some minor contributors. I do not think that it's a problem. I would expect that if we do not have a disagreement, it is fine. The question is, if we need automake/autoconf tools to package those files. It seems to me that it might be clever to keep this ASIS for now. Users/packagers can just do 'cp' to the appropriate directory. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ASoC: samsung: i2s: Add clocks' macros descriptions
by Marek Szyprowski 24 Sep '19

24 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> To increase macro readability added descriptions to clocks macros. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- include/dt-bindings/sound/samsung-i2s.h | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/include/dt-bindings/sound/samsung-i2s.h b/include/dt-bindings/sound/samsung-i2s.h index 77545f14c379..fd752475c762 100644 --- a/include/dt-bindings/sound/samsung-i2s.h +++ b/include/dt-bindings/sound/samsung-i2s.h @@ -2,8 +2,13 @@ #ifndef _DT_BINDINGS_SAMSUNG_I2S_H #define _DT_BINDINGS_SAMSUNG_I2S_H -#define CLK_I2S_CDCLK 0 -#define CLK_I2S_RCLK_SRC 1 -#define CLK_I2S_RCLK_PSR 2 +/* the CDCLK (CODECLKO) gate clock */ +#define CLK_I2S_CDCLK 0 + +/* the RCLKSRC mux clock (corresponding to RCLKSRC bit in IISMOD register) */ +#define CLK_I2S_RCLK_SRC 1 + +/* the RCLK prescaler divider clock (corresponding to the IISPSR register) */ +#define CLK_I2S_RCLK_PSR 2 #endif /* _DT_BINDINGS_SAMSUNG_I2S_H */ -- 2.17.1

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[alsa-devel] Propagating audio properties along the audio path
by Marc Gonzalez 24 Sep '19

24 Sep '19

Hello everyone, Disclaimer: I've never worked in the sound/ layer, and it is possible that some of my questions are silly or obvious. Basically, I'm trying to implement some form of eARC(*) on an arm64 SoC. (*) enhanced Audio Return Channel (from HDMI 2.1) The setup looks like this: A = Some kind of audio source, typically a TV or game console B = The arm64 SoC, equipped with some nice speakers HDMI A ------> B If we look inside B, we actually have B1 = an eARC receiver (input = HDMI, output = I2S) B2 = an audio DSP (input = I2S, output = speakers) I2S ? B1 -----> B2 -----> speakers If I read the standard right, B is supposed to advertise which audio formats it supports, and A is supposed to pick "the best". For the sake of argument, let's say A picks "PCM, 48 kHz, 8 channels, 16b". At some point, B receives audio packets, parses the Channel Status, and determines that A is sending "PCM, 48 kHz, 8 channels, 16b". The driver then configures the I2S link, and forwards the audio stream over I2S to the DSP. QUESTION_1: How is the DSP supposed to "learn" the properties of the audio stream? (AFAIU, they're not embedded in the data, so there must be some side-channel?) I assume the driver of B1 is supposed to propagate the info to the driver of B2? (Via some call-backs? By calling a function in B2?) QUESTION_2: Does it ever make sense for B2 to ask B1 to change the audio properties? (Not sure if B1 is even allowed to renegotiate.) Regards.

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[alsa-devel] [PATCH 0/2] Add CML M/C for RT5682 and RT1011
by sathya.prakash.m.r＠intel.com 24 Sep '19

24 Sep '19

From: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Naveen Manohar (2): ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 ASoC: Intel: Add acpi match for rt1011 based m/c driver sound/soc/intel/boards/Kconfig | 18 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 475 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 502 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- V1: Acked-by:pierre-louis.bossart@linux.intel.com at https://github.com/thesofproject/linux/pull/1161 1.9.1

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[alsa-devel] [PATCH v4] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Marek Szyprowski 24 Sep '19

24 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung I2S controller to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v4: - Removed description of i2s device nodes' clocks from 'clocks' property. - Added 'clock-output-names' property. - Added description of clock names. - Added '#clock-cells' property to required properties - Description of the provided clocks moved to samsung-i2s.h in separate patch Best regards, Maciej Falkowski --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ----------- .../bindings/sound/samsung-i2s.yaml | 136 ++++++++++++++++++ 2 files changed, 136 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..7d2750c26f11 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,136 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with + secondary fifo, s/w reset control and internal mux for root clk src. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for + playback, stereo channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + + samsung,exynos7-i2s: with all the available features of exynos5 i2s. + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for i2s0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for i2s1 and i2s2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clk. i2s0 has internal mux to select the source + of root clk and i2s1 and i2s2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: # deprecated + oneOf: + - items: # for i2s0 + - const: i2s_cdclk0 + - items: # for i2s1 + - const: i2s_cdclk1 + - items: # for i2s2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + sub system(used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + - "#clock-cells" + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; + -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add DSD support for EVGA NU Audio
by Jussi Laako 24 Sep '19

24 Sep '19

EVGA NU Audio is actually a USB audio device on a PCIexpress card, with it's own USB controller. It supports both PCM and DSD. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5fd4ccce452d..fbfde996fee7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1658,6 +1658,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ case 0x2ab6: /* T+A devices */ + case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; -- 2.17.1

2 1

[alsa-devel] [PATCH alsa-lib] pcm: chmap: Fix memory leak at snd_pcm_set_chmap()
by Takashi Iwai 24 Sep '19

24 Sep '19

snd_pcm_set_chmap() leaks the memory returned from snd_pcm_get_chmap() without releasing. Add the missing free() call as well as a slight code refactoring. Reported-by: Conrad Jones BugLink: https://github.com/alsa-project/alsa-lib/pull/11 Fixes: d20e24e5d161 ("chmap: Always succeed setting the map to what it already is") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 178d43875f00..d6bf31ac7ae3 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -8099,8 +8099,13 @@ snd_pcm_chmap_t *snd_pcm_get_chmap(snd_pcm_t *pcm) */ int snd_pcm_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map) { - const snd_pcm_chmap_t *oldmap = snd_pcm_get_chmap(pcm); - if (oldmap && chmap_equal(oldmap, map)) + const snd_pcm_chmap_t *oldmap; + int nochange; + + oldmap = snd_pcm_get_chmap(pcm); + nochange = (oldmap && chmap_equal(oldmap, map)); + free((void *)oldmap); + if (nochange) return 0; if (!pcm->ops->set_chmap) -- 2.16.4

1 0

[alsa-devel] TDM audio support and sample size updates for tegra30
by Ben Dooks 24 Sep '19

24 Sep '19

A patch series for TDM audio, non 16bit sample size and a small fifo fix for tegra30 series SoC. Changes since v1: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock

3 11

Re: [alsa-devel] [PATCH V3 4/4] ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8
by S.j. Wang 24 Sep '19

24 Sep '19

Hi > > One issue for error-out and some nit-pickings inline. Thanks. > > On Thu, Sep 19, 2019 at 08:11:42PM +0800, Shengjiu Wang wrote: > > There is error "aplay: pcm_write:2023: write error: Input/output error" > > on i.MX8QM/i.MX8QXP platform for S24_3LE format. > > > > In i.MX8QM/i.MX8QXP, the DMA is EDMA, which don't support 24bit > > sample, but we didn't add any constraint, that cause issues. > > > > So we need to query the caps of dma, then update the hw parameters > > according to the caps. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/fsl_asrc.c | 4 +-- > > sound/soc/fsl/fsl_asrc.h | 3 +++ > > sound/soc/fsl/fsl_asrc_dma.c | 52 > > +++++++++++++++++++++++++++++++----- > > 3 files changed, 50 insertions(+), 9 deletions(-) > > > > @@ -276,6 +274,11 @@ static int fsl_asrc_dma_startup(struct > snd_pcm_substream *substream) > > struct device *dev = component->dev; > > struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); > > struct fsl_asrc_pair *pair; > > + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; > > + u8 dir = tx ? OUT : IN; > > + struct dma_chan *tmp_chan; > > + struct snd_dmaengine_dai_dma_data *dma_data; > > Nit: would it be possible to reorganize these a bit? Usually we put struct > things together unless there is a dependency, similar to > fsl_asrc_dma_hw_params(). > > > @@ -285,9 +288,44 @@ static int fsl_asrc_dma_startup(struct > > snd_pcm_substream *substream) > > > > runtime->private_data = pair; > > > > + /* Request a temp pair, which is release in the end */ > > Nit: "which will be released later" or "and will release it later"? And could > we use a work like "dummy"? Or at least I would love to see the comments > explaining the parameter "1" > in the function call below. > > > + ret = fsl_asrc_request_pair(1, pair); > > + if (ret < 0) { > > + dev_err(dev, "failed to request asrc pair\n"); > > + return ret; > > + } > > + > > + tmp_chan = fsl_asrc_get_dma_channel(pair, dir); > > + if (!tmp_chan) { > > + dev_err(dev, "can't get dma channel\n"); > > Could we align with other error messages using "failed to"? > > > + ret = snd_soc_set_runtime_hwparams(substream, > &snd_imx_hardware); > > + if (ret) > > + return ret; > > + > [...] > > + dma_release_channel(tmp_chan); > > + fsl_asrc_release_pair(pair); > > I think we need an "out:" here for those error-out routines to goto. > Otherwise, it'd be a pair leak? > > > + > > Could we drop this? There is a blank line below already :) > Will update them. Best regards Wang Shengjiu

1 0

[alsa-devel] [PATCH] ASoC: soc-component: fix a couple missing error assignments
by Dan Carpenter 24 Sep '19

24 Sep '19

There were a couple places where the return value wasn't assigned so the error handling wouldn't trigger. Fixes: 5c0769af4caf ("ASoC: soc-dai: add snd_soc_dai_bespoke_trigger()") Fixes: 95aef3553384 ("ASoC: soc-dai: add snd_soc_dai_trigger()") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..a1b99ac57d9e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1070,7 +1070,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - snd_soc_dai_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; @@ -1097,7 +1097,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, return ret; } - snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; -- 2.20.1

2 1

[alsa-devel] [PATCH trivial] sound: Fix Kconfig indentation
by Krzysztof Kozlowski 23 Sep '19

23 Sep '19

Adjust indentation from spaces to tab (+optional two spaces) as in coding style with command like: $ sed -e 's/^ /\t/' -i */Kconfig Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/core/Kconfig | 28 ++++++++++---------- sound/drivers/Kconfig | 20 +++++++------- sound/firewire/Kconfig | 6 ++--- sound/isa/Kconfig | 18 ++++++------- sound/mips/Kconfig | 12 ++++----- sound/pci/Kconfig | 2 +- sound/soc/cirrus/Kconfig | 14 +++++----- sound/soc/codecs/Kconfig | 32 +++++++++++------------ sound/soc/intel/Kconfig | 2 +- sound/soc/intel/boards/Kconfig | 48 +++++++++++++++++----------------- sound/soc/pxa/Kconfig | 16 ++++++------ sound/soc/qcom/Kconfig | 20 +++++++------- sound/soc/samsung/Kconfig | 8 +++--- sound/soc/sof/imx/Kconfig | 12 ++++----- sound/soc/sof/intel/Kconfig | 22 ++++++++-------- sound/soc/xilinx/Kconfig | 20 +++++++------- sound/soc/zte/Kconfig | 12 ++++----- sound/usb/Kconfig | 32 +++++++++++------------ 18 files changed, 162 insertions(+), 162 deletions(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 4ee79ad6ae22..4044c42d8595 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -72,11 +72,11 @@ config SND_PCM_OSS config SND_PCM_OSS_PLUGINS bool "OSS PCM (digital audio) API - Include plugin system" depends on SND_PCM_OSS - default y + default y help - If you disable this option, the ALSA's OSS PCM API will not - support conversion of channels, formats and rates. It will - behave like most of new OSS/Free drivers in 2.4/2.6 kernels. + If you disable this option, the ALSA's OSS PCM API will not + support conversion of channels, formats and rates. It will + behave like most of new OSS/Free drivers in 2.4/2.6 kernels. config SND_PCM_TIMER bool "PCM timer interface" if EXPERT @@ -128,13 +128,13 @@ config SND_SUPPORT_OLD_API or older). config SND_PROC_FS - bool "Sound Proc FS Support" if EXPERT - depends on PROC_FS - default y - help - Say 'N' to disable Sound proc FS, which may reduce code size about - 9KB on x86_64 platform. - If unsure say Y. + bool "Sound Proc FS Support" if EXPERT + depends on PROC_FS + default y + help + Say 'N' to disable Sound proc FS, which may reduce code size about + 9KB on x86_64 platform. + If unsure say Y. config SND_VERBOSE_PROCFS bool "Verbose procfs contents" @@ -142,8 +142,8 @@ config SND_VERBOSE_PROCFS default y help Say Y here to include code for verbose procfs contents (provides - useful information to developers when a problem occurs). On the - other side, it makes the ALSA subsystem larger. + useful information to developers when a problem occurs). On the + other side, it makes the ALSA subsystem larger. config SND_VERBOSE_PRINTK bool "Verbose printk" @@ -164,7 +164,7 @@ config SND_DEBUG_VERBOSE depends on SND_DEBUG help Say Y here to enable extra-verbose debugging messages. - + Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages. So, say Y only if you are ready to be annoyed. diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 09932cc98e9d..15d6d46acf9c 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_MPU401_UART - tristate - select SND_RAWMIDI + tristate + select SND_RAWMIDI config SND_OPL3_LIB tristate @@ -90,16 +90,16 @@ config SND_DUMMY will be called snd-dummy. config SND_ALOOP - tristate "Generic loopback driver (PCM)" - select SND_PCM - help - Say 'Y' or 'M' to include support for the PCM loopback device. + tristate "Generic loopback driver (PCM)" + select SND_PCM + help + Say 'Y' or 'M' to include support for the PCM loopback device. This module returns played samples back to the user space using the standard ALSA PCM device. The devices are routed 0->1 and - 1->0, where first number is the playback PCM device and second + 1->0, where first number is the playback PCM device and second number is the capture device. Module creates two PCM devices and configured number of substreams (see the pcm_substreams module - parameter). + parameter). The loopback device allows time sychronization with an external timing source using the time shift universal control (+-20% @@ -142,12 +142,12 @@ config SND_MTS64 select SND_RAWMIDI help The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with - additional SMPTE Timecode capabilities for the parallel port. + additional SMPTE Timecode capabilities for the parallel port. Say 'Y' to include support for this device. To compile this driver as a module, chose 'M' here: the module - will be called snd-mts64. + will be called snd-mts64. config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index b0a904cdb932..995c2cefc222 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -77,7 +77,7 @@ config SND_BEBOB tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" select SND_FIREWIRE_LIB select SND_HWDEP - help + help Say Y here to include support for FireWire devices based on BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware: * Edirol FA-66/FA-101 @@ -111,8 +111,8 @@ config SND_BEBOB * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro - To compile this driver as a module, choose M here: the module - will be called snd-bebob. + To compile this driver as a module, choose M here: the module + will be called snd-bebob. config SND_FIREWIRE_DIGI00X tristate "Digidesign Digi 002/003 family support" diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b690ed937cbe..6ffa48dd5983 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -2,22 +2,22 @@ # ALSA ISA drivers config SND_WSS_LIB - tristate - select SND_PCM + tristate + select SND_PCM select SND_TIMER config SND_SB_COMMON - tristate + tristate config SND_SB8_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON config SND_SB16_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON menuconfig SND_ISA bool "ISA sound devices" diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 8a33402fd415..b497b803c834 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -14,15 +14,15 @@ config SND_SGI_O2 tristate "SGI O2 Audio" depends on SGI_IP32 select SND_PCM - help - Sound support for the SGI O2 Workstation. + help + Sound support for the SGI O2 Workstation. config SND_SGI_HAL2 - tristate "SGI HAL2 Audio" - depends on SGI_HAS_HAL2 + tristate "SGI HAL2 Audio" + depends on SGI_HAS_HAL2 select SND_PCM - help - Sound support for the SGI Indy and Indigo2 Workstation. + help + Sound support for the SGI Indy and Indigo2 Workstation. endif # SND_MIPS diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7630f808d087..93bc9bef7641 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -217,7 +217,7 @@ config SND_CMIPCI will be called snd-cmipci. config SND_OXYGEN_LIB - tristate + tristate config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 2333efac758a..8039a8febefa 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -33,13 +33,13 @@ config SND_EP93XX_SOC_AC97 select SND_SOC_AC97_BUS config SND_EP93XX_SOC_SNAPPERCL15 - tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C - select SND_EP93XX_SOC_I2S - select SND_SOC_TLV320AIC23_I2C - help - Say Y or M here if you want to add support for I2S audio on the - Bluewater Systems Snapper CL15 module. + tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C + select SND_EP93XX_SOC_I2S + select SND_SOC_TLV320AIC23_I2C + help + Say Y or M here if you want to add support for I2S audio on the + Bluewater Systems Snapper CL15 module. config SND_EP93XX_SOC_SIMONE tristate "SoC Audio support for Simplemachines Sim.One board" diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89f8c5a..ef9d73b89623 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -257,16 +257,16 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - help - Normally ASoC codec drivers are only built if a machine driver which - uses them is also built since they are only usable with a machine - driver. Selecting this option will allow these drivers to be built - without an explicit machine driver for test and development purposes. + help + Normally ASoC codec drivers are only built if a machine driver which + uses them is also built since they are only usable with a machine + driver. Selecting this option will allow these drivers to be built + without an explicit machine driver for test and development purposes. Support for the bus types used to access the codecs to be built must be selected separately. - If unsure select "N". + If unsure select "N". config SND_SOC_88PM860X tristate @@ -570,8 +570,8 @@ config SND_SOC_CS42XX8_I2C # Cirrus Logic CS43130 HiFi DAC config SND_SOC_CS43130 - tristate "Cirrus Logic CS43130 CODEC" - depends on I2C + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C config SND_SOC_CS4341 tristate "Cirrus Logic CS4341 CODEC" @@ -643,19 +643,19 @@ config SND_SOC_L3 tristate config SND_SOC_DA7210 - tristate + tristate config SND_SOC_DA7213 - tristate + tristate config SND_SOC_DA7218 tristate config SND_SOC_DA7219 - tristate + tristate config SND_SOC_DA732X - tristate + tristate config SND_SOC_DA9055 tristate @@ -717,7 +717,7 @@ config SND_SOC_INNO_RK3036 select REGMAP_MMIO config SND_SOC_ISABELLE - tristate + tristate config SND_SOC_LM49453 tristate @@ -988,7 +988,7 @@ config SND_SOC_RT5640 tristate config SND_SOC_RT5645 - tristate + tristate config SND_SOC_RT5651 tristate @@ -1220,7 +1220,7 @@ config SND_SOC_UDA134X tristate config SND_SOC_UDA1380 - tristate + tristate depends on I2C config SND_SOC_WCD9335 @@ -1348,7 +1348,7 @@ config SND_SOC_WM8904 depends on I2C config SND_SOC_WM8940 - tristate + tristate config SND_SOC_WM8955 tristate diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..597516c0c192 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -113,7 +113,7 @@ config SND_SOC_INTEL_SKYLAKE select SND_SOC_INTEL_CNL select SND_SOC_INTEL_CFL help - This is a backwards-compatible option to select all devices + This is a backwards-compatible option to select all devices supported by the Intel SST/Skylake driver. This option is no longer recommended and will be deprecated when the SOF driver is introduced. Distributions should explicitly diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7ab4a5f..882ff36a7c9c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -3,13 +3,13 @@ menuconfig SND_SOC_INTEL_MACH bool "Intel Machine drivers" depends on SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL help - Intel ASoC Machine Drivers. If you have a Intel machine that - has an audio controller with a DSP and I2S or DMIC port, then - enable this option by saying Y + Intel ASoC Machine Drivers. If you have a Intel machine that + has an audio controller with a DSP and I2S or DMIC port, then + enable this option by saying Y - Note that the answer to this question doesn't directly affect the - kernel: saying N will just cause the configurator to skip all - the questions about Intel ASoC machine drivers. + Note that the answer to this question doesn't directly affect the + kernel: saying N will just cause the configurator to skip all + the questions about Intel ASoC machine drivers. if SND_SOC_INTEL_MACH @@ -114,11 +114,11 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5670 - help - This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell - platforms with RT5672 audio codec. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" @@ -311,20 +311,20 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH If unsure select "N". config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH - tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" + tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SPI - select SND_SOC_RT5663 - select SND_SOC_RT5514 - select SND_SOC_RT5514_SPI - select SND_SOC_MAX98927 - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for RT5663 + RT5514 + MAX98927. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + depends on SPI + select SND_SOC_RT5663 + select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI + select SND_SOC_MAX98927 + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for RT5663 + RT5514 + MAX98927. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" @@ -393,7 +393,7 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH help This adds support for ASoC machine driver for Intel platforms SKL/KBL/BXT/APL with iDisp, HDA audio codecs. - Say Y or m if you have such a device. This is a recommended option. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 213d4dab0346..295cfffa4646 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -190,14 +190,14 @@ config SND_PXA2XX_SOC_MAGICIAN HTC Magician. config SND_PXA2XX_SOC_MIOA701 - tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 depends on AC97_BUS=n - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9713 - help - Say Y if you want to add support for SoC audio on the - MIO A701. + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" @@ -205,7 +205,7 @@ config SND_PXA2XX_SOC_IMOTE2 select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help - Say Y if you want to add support for SoC audio on the + Say Y if you want to add support for SoC audio on the IMote 2. config SND_MMP_SOC_BROWNSTONE diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 60086858e920..6530d2462a9e 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -3,8 +3,8 @@ config SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST help - Say Y or M if you want to add support to use audio devices - in Qualcomm Technologies SOC-based platforms. + Say Y or M if you want to add support to use audio devices + in Qualcomm Technologies SOC-based platforms. config SND_SOC_LPASS_CPU tristate @@ -30,17 +30,17 @@ config SND_SOC_STORM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help - Say Y or M if you want add support for SoC audio on the - Qualcomm Technologies IPQ806X-based Storm board. + Say Y or M if you want add support for SoC audio on the + Qualcomm Technologies IPQ806X-based Storm board. config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 help - Support for Qualcomm Technologies LPASS audio block in - APQ8016 SOC-based systems. - Say Y if you want to use audio devices on MI2S. + Support for Qualcomm Technologies LPASS audio block in + APQ8016 SOC-based systems. + Say Y if you want to use audio devices on MI2S. config SND_SOC_QCOM_COMMON tristate @@ -93,9 +93,9 @@ config SND_SOC_MSM8996 select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON help - Support for Qualcomm Technologies LPASS audio block in - APQ8096 SoC-based systems. - Say Y if you want to use audio device on this SoCs + Support for Qualcomm Technologies LPASS audio block in + APQ8096 SoC-based systems. + Say Y if you want to use audio device on this SoCs config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 638983123d8f..9304177de78a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -195,10 +195,10 @@ config SND_SOC_ODROID Say Y here to enable audio support for the Odroid XU3/XU4. config SND_SOC_ARNDALE_RT5631_ALC5631 - tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on I2C - select SND_SAMSUNG_I2S - select SND_SOC_RT5631 + tristate "Audio support for RT5631(ALC5631) on Arndale Board" + depends on I2C + select SND_SAMSUNG_I2S + select SND_SOC_RT5631 config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750..6315fba8ce71 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -5,9 +5,9 @@ config SND_SOC_SOF_IMX_TOPLEVEL depends on ARM64|| COMPILE_TEST depends on SND_SOC_SOF_OF help - This adds support for Sound Open Firmware for NXP i.MX platforms. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX platforms. + Say Y if you have such a device. + If unsure select "N". if SND_SOC_SOF_IMX_TOPLEVEL @@ -16,8 +16,8 @@ config SND_SOC_SOF_IMX8 depends on IMX_SCU depends on IMX_DSP help - This adds support for Sound Open Firmware for NXP i.MX8 platforms - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX8 platforms + Say Y if you have such a device. + If unsure select "N". endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..18c41fdc47c8 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -36,7 +36,7 @@ config SND_SOC_SOF_INTEL_PCI config SND_SOC_SOF_INTEL_HIFI_EP_IPC tristate help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_INTEL_ATOM_HIFI_EP @@ -217,31 +217,31 @@ config SND_SOC_SOF_COMETLAKE_H_SUPPORT config SND_SOC_SOF_TIGERLAKE_SUPPORT bool "SOF support for Tigerlake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the Tigerlake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the Tigerlake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_TIGERLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_ELKHARTLAKE_SUPPORT bool "SOF support for ElkhartLake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the ElkhartLake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the ElkhartLake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_ELKHARTLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_HDA_COMMON diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 69973179ef15..1d3586b68db7 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,15 +9,15 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" - help - Select this option to enable Xilinx audio formatter - support. This provides DMA platform device support for - audio functionality. + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" - help - Select this option to enable Xilinx SPDIF Audio. - This provides playback and capture of SPDIF audio in - AES format. + tristate "Audio support for the the Xilinx SPDIF" + help + Select this option to enable Xilinx SPDIF Audio. + This provides playback and capture of SPDIF audio in + AES format. diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig index a7842e4b791c..a23d4f13ca19 100644 --- a/sound/soc/zte/Kconfig +++ b/sound/soc/zte/Kconfig @@ -18,9 +18,9 @@ config ZX_I2S ZTE ZX I2S interface config ZX_TDM - tristate "ZTE ZX TDM Driver Support" - depends on COMMON_CLK - select SND_SOC_GENERIC_DMAENGINE_PCM - help - Say Y or M if you want to add support for codecs attached to the - ZTE ZX TDM interface + tristate "ZTE ZX TDM Driver Support" + depends on COMMON_CLK + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for codecs attached to the + ZTE ZX TDM interface diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e2c53a0841da..059242f15d75 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -107,24 +107,24 @@ config SND_USB_US122L will be called snd-usb-us122l. config SND_USB_6FIRE - tristate "TerraTec DMX 6Fire USB" - select FW_LOADER - select BITREVERSE - select SND_RAWMIDI - select SND_PCM - select SND_VMASTER - help - Say Y here to include support for TerraTec 6fire DMX USB interface. - - You will need firmware files in order to be able to use the device - after it has been coldstarted. An install script for the firmware - and further help can be found at - http://sixfireusb.sourceforge.net + tristate "TerraTec DMX 6Fire USB" + select FW_LOADER + select BITREVERSE + select SND_RAWMIDI + select SND_PCM + select SND_VMASTER + help + Say Y here to include support for TerraTec 6fire DMX USB interface. + + You will need firmware files in order to be able to use the device + after it has been coldstarted. An install script for the firmware + and further help can be found at + http://sixfireusb.sourceforge.net config SND_USB_HIFACE - tristate "M2Tech hiFace USB-SPDIF driver" - select SND_PCM - help + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help Select this option to include support for M2Tech hiFace USB-SPDIF interface. -- 2.17.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v5.4
by Mark Brown 23 Sep '19

23 Sep '19

The following changes since commit fca11622d600228bec405456f41590155b3a3eca: ASoC: sdm845: remove unneeded semicolon (2019-09-15 10:29:51 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.4-rc1 for you to fetch changes up to 147162f575152db80000fb3de26358264768ee9f: ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies (2019-09-20 12:06:48 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.4 A small smattering of ASoC fixes for v5.4 - nothing too exciting here, all small standalone things. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies Bard Liao (1): ASoC: core: delete component->card_list in soc_remove_component only Bard liao (1): ASoC: core: use list_del_init and move it back to soc_cleanup_component Gregory CLEMENT (1): ASoC: atmel_ssc_dai: Remove wrong spinlock usage Kuninori Morimoto (1): ASoC: rsnd: do error check after rsnd_channel_normalization() Mihai Serban (1): ASoC: fsl_sai: Fix noise when using EDMA Peter Ujfalusi (1): ASoC: pcm3168a: The codec does not support S32_LE sound/soc/atmel/atmel_ssc_dai.c | 12 ++---------- sound/soc/atmel/atmel_ssc_dai.h | 1 - sound/soc/codecs/pcm3168a.c | 3 +-- sound/soc/fsl/fsl_sai.c | 15 +++++++++++++++ sound/soc/fsl/fsl_sai.h | 1 + sound/soc/sh/rcar/ssi.c | 10 +++++----- sound/soc/soc-core.c | 2 +- sound/soc/ti/Kconfig | 11 +++++++++-- 8 files changed, 34 insertions(+), 21 deletions(-)

2 1

[alsa-devel] [PATCH V3 0/4] update supported sample format
by Shengjiu Wang 21 Sep '19

21 Sep '19

This patch serial is to update the supported format for fsl_asrc and fix some format issue. Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch changes in v3 - Fix build report by kbuild test robot <lkp(a)intel.com> - change snd_dmaengine_pcm_set_runtime_hwparams to snd_dmaengine_pcm_refine_runtime_hwparams include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 52 ++++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 62 +++----------------- 6 files changed, 188 insertions(+), 86 deletions(-) -- 2.21.0

2 7

[alsa-devel] [PATCH 1/2] PCI: Add a helper to check Power Resource Requirements _PR3 existence
by Kai-Heng Feng 20 Sep '19

20 Sep '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> --- drivers/pci/pci.c | 20 ++++++++++++++++++++ include/linux/pci.h | 2 ++ 2 files changed, 22 insertions(+) diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index 1b27b5af3d55..776af15b92c2 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -5856,6 +5856,26 @@ int pci_set_vga_state(struct pci_dev *dev, bool decode, return 0; } +bool pci_pr3_present(struct pci_dev *pdev) +{ + struct pci_dev *parent_pdev = pci_upstream_bridge(pdev); + struct acpi_device *parent_adev; + + if (acpi_disabled) + return false; + + if (!parent_pdev) + return false; + + parent_adev = ACPI_COMPANION(&parent_pdev->dev); + if (!parent_adev) + return false; + + return parent_adev->power.flags.power_resources && + acpi_has_method(parent_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 82e4cd1b7ac3..9b6f7b67fac9 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2348,9 +2348,11 @@ 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

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[alsa-devel] [PATCH v7 0/4] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 20 Sep '19

20 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. The dependent change on hdmi-codec.c https://patchwork.kernel.org/patch/11047447 has been picked up by Mark Brown in ASoC tree for-5.4 branch. Changes from v6 to v7: 1. rockchip_max98090: Use phandle of HDMI from DTS to find codec_dai. With this we don't need to set fixed id for the created hdmi-audio-codec device. Cheng-Yi Chiang (4): drm: bridge: dw-hdmi: Report connector status using callback ASoC: rockchip-max98090: Add description for rockchip,hdmi-codec ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../bindings/sound/rockchip-max98090.txt | 2 + .../boot/dts/rk3288-veyron-analog-audio.dtsi | 1 + .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 11 ++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++- include/drm/bridge/dw_hdmi.h | 4 + sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rockchip_max98090.c | 149 ++++++++++++++---- 7 files changed, 182 insertions(+), 29 deletions(-) -- 2.23.0.237.gc6a4ce50a0-goog

2 6

[alsa-devel] [PATCH] ASoC: core: use list_del_init and move it back to soc_cleanup_component
by Bard liao 20 Sep '19

20 Sep '19

commit a0a4bf57a977 ("ASoC: core: delete component->card_list in soc_remove_component only") was trying to fix a kernel oops when list_del was called twice without re-init the list. Use list_del_init() can solve it, too. Besides, it will be more readable if we cleanup all component related resource at soc_cleanup_component(). Suggested-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index aff4b4bf4d07..88978a3036c4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -978,6 +978,7 @@ static void soc_cleanup_component(struct snd_soc_component *component) /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); + list_del_init(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; @@ -990,7 +991,7 @@ static void soc_remove_component(struct snd_soc_component *component) return; snd_soc_component_remove(component); - list_del(&component->card_list); + soc_cleanup_component(component); } -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: pcm3168a: The codec does not support S32_LE
by Peter Ujfalusi 20 Sep '19

20 Sep '19

24 bits is supported in all modes and 16 bit only when the codec is slave and the DAI is set to RIGHT_J. Remove the unsupported sample format. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 50ed86d45c26..88b75695fbf7 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -21,8 +21,7 @@ #define PCM3168A_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S24_LE) #define PCM3168A_FMT_I2S 0x0 #define PCM3168A_FMT_LEFT_J 0x1 -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies
by Arnd Bergmann 20 Sep '19

20 Sep '19

SND_SOC_DM365_VOICE_CODEC is a 'bool' option in a choice statement, meaning it cannot be set to =m, but it selects two other drivers that we may want to be loadable modules after all: WARNING: unmet direct dependencies detected for SND_SOC_CQ0093VC Depends on [m]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] Selected by [y]: - SND_SOC_DM365_VOICE_CODEC [=y] && <choice> Selected by [m]: - SND_SOC_ALL_CODECS [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMPILE_TEST [=y] Add an intermediate symbol that sets SND_SOC_CQ0093VC and MFD_DAVINCI_VOICECODEC to =m if SND_SOC=m. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/ti/Kconfig | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 87a9b9dd4e98..29f61053ab62 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -200,11 +200,18 @@ config SND_SOC_DM365_AIC3X_CODEC config SND_SOC_DM365_VOICE_CODEC bool "Voice Codec - CQ93VC" - select MFD_DAVINCI_VOICECODEC - select SND_SOC_CQ0093VC help Say Y if you want to add support for SoC On-chip voice codec endchoice +config SND_SOC_DM365_VOICE_CODEC_MODULE + def_tristate y + depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC + select MFD_DAVINCI_VOICECODEC + select SND_SOC_CQ0093VC + help + The is an internal symbol needed to ensure that the codec + and MFD driver can be built as loadable modules if necessary. + endmenu -- 2.20.0

2 1

[alsa-devel] [PATCH] ASoC: soc-component.h: tidyup for soc-component.h
by Kuninori Morimoto 20 Sep '19

20 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-component.h already has SPDX License, GPL explanation is not needed. And it is included from soc.h only. No need to header-test. This patch tidyup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark this is for for-5.4 include/Kbuild | 1 + include/sound/soc-component.h | 6 ------ 2 files changed, 1 insertion(+), 6 deletions(-) diff --git a/include/Kbuild b/include/Kbuild index c38f0d4..c6909b5 100644 --- a/include/Kbuild +++ b/include/Kbuild @@ -1052,6 +1052,7 @@ header-test- += sound/soc-acpi-intel-match.h header-test- += sound/soc-dai.h header-test- += sound/soc-dapm.h header-test- += sound/soc-dpcm.h +header-test- += sound/soc-component.h header-test- += sound/sof/control.h header-test- += sound/sof/dai-intel.h header-test- += sound/sof/dai.h diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2e..a869199 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -3,16 +3,10 @@ * soc-component.h * * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SOC_COMPONENT_H #define __SOC_COMPONENT_H -#include <sound/soc.h> - /* * Component probe and remove ordering levels for components with runtime * dependencies. -- 2.7.4

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[alsa-devel] RfC / [PATCH] Jack sense support for Medion E4254
by Jan-Marek Glogowski 20 Sep '19

20 Sep '19

Hi everybody, I recently bought the laptop. It has one jack for attaching a headset. Jack sense is announced by the HDA config but not working. Except for some positioning, and color information, the PIN config seemed to be sensible, and everything I tried to override, didn't result in a working jack. Manually unmuting the jack using hda-analyser resulted in working output, so generally the config seemed to work. Reading the patch-realtek code, if found ALC256_FIXUP_ASUS_HEADSET_MODE, which somehow changed the pin 0x19 to be associated with the 0x21 headset out pin, resulting in this dmesg change: -snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 +snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 And for whatever reason jack sense is now working, at least for some normal output-only headset (currently I don't have one included a mic with me, so actually can't test that mic volume works correctly). I was just wondering, if my pin values are actually correct, because they still define different association, as I understand it, but the ASUS fix even works with these wrong channels /associations 2 and 4. So the attached patch works for me, but I'm still wondering, if this is the correct solution. FWIW, jack sense works fine in Windows. Anything I might be missing here? Jan-Marek ------- >From 0ccc5f07d8737690bd2df6d88a5af0f1f36d0e40 Mon Sep 17 00:00:00 2001 From: Jan-Marek Glogowski <glogow(a)fbihome.de> Date: Sun, 15 Sep 2019 16:57:28 +0200 Subject: [PATCH] ALSA: hda/realtek: PCI quirk for Medion E4254 The laptop has a combined jack to attach headsets on the right. The BIOS encodes them as two different colored jacks at the front, but otherwise it seems to be configured ok. But any adaption of the pins config on its own doesn't fix the jack detection to work in Linux. Still Windows works correct. This is somehow fixed by chaining ALC256_FIXUP_ASUS_HEADSET_MODE, which seems to register the microphone jack as a headset part and also results in fixing jack sensing, visible in dmesg as: -snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 +snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 Signed-off-by: Jan-Marek Glogowski <glogow(a)fbihome.de> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1ddfd2fac52..448e0146dab2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5800,6 +5800,7 @@ enum { ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, + ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6850,6 +6851,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a11040 }, + { 0x21, 0x04211020 }, + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7113,6 +7124,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. @@ -7280,6 +7292,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"}, {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"}, {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, + {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.20.1

2 3

[alsa-devel] [PATCH] ALSA: hda - Apply AMD controller workaround for Raven platform
by Takashi Iwai 20 Sep '19

20 Sep '19

It's reported that the garbled sound on HP Envy x360 13z-ag000 (Ryzen Laptop) is fixed by the same workaround applied to other AMD chips. Update the driver_data entry for Raven (1022:15e3) to use the newly introduced preset, AZX_DCAPS_PRESET_AMD_SB. Since it already contains AZX_DCAPS_PM_RUNTIME, we can drop that bit, too. Reported-and-tested-by: Dennis Padiernos <depadiernos(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 91e71be42fa4..240f4ca76391 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2485,8 +2485,7 @@ static const struct pci_device_id azx_ids[] = { AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | - AZX_DCAPS_PM_RUNTIME }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, -- 2.16.4

1 0

[alsa-devel] alsa-lib: Add ucm support for whiskeylake sof-skl_hda_card audio
by Elimar Riesebieter 19 Sep '19

19 Sep '19

Hi ALSA devs, please notice the attached suggestion from Lenovo to support sound for whiskeylake sof-skl_hda_card audio. Thanks Elimar -- You cannot propel yourself forward by patting yourself on the back.

4 3

[alsa-devel] [PATCH v1 0/9] Basic sound support for Arndale board / wm8994 updates
by Sylwester Nawrocki 19 Sep '19

19 Sep '19

This patch series adds basic audio support for Exynos5250 SoC based Arndale board, the Bluetooth receiver source and HDMI output are not covered yet. There is also one fix for wm8994 driver related to WM1811 CODEC and wm8994 updates to handle MCLK clocks, similar to patches: ae1ea48c5c59 ("ASoC: arizona: Add gating for source clocks of the FLLs") 7a4413d0dc96 ("ASoC: arizona: Add gating for clock when used for direct MCLK") Sylwester Nawrocki (9): ASoC: wm8994: Do not register inapplicable controls for WM1811 mfd: wm8994: Add support for MCLKn clock control ASoC: wm8994: Add support for setting MCLKn clock rate ASoC: wm8994: Add support for MCLKn clock gating ASoC: samsung: arndale: Simplify DAI link initialization ASoC: dt-bindings: Document "samsung,arndale-wm1811" compatible ASoC: samsung: arndale: Add support for WM1811 CODEC ASoC: samsung: arndale: Add missing OF node dereferencing ARM: dts: arndale: Add audio support (WM1811 CODEC boards) .../devicetree/bindings/sound/arndale.txt | 5 +- arch/arm/boot/dts/exynos5250-arndale.dts | 27 ++- drivers/mfd/wm8994-core.c | 9 + include/linux/mfd/wm8994/core.h | 9 + sound/soc/codecs/wm8994.c | 164 +++++++++++++++--- sound/soc/samsung/arndale_rt5631.c | 155 +++++++++++++---- 6 files changed, 306 insertions(+), 63 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v6] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 19 Sep '19

19 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

2 2

[alsa-devel] DPCM: suspicious code in dpcm_prune_paths()
by Guennadi Liakhovetski 19 Sep '19

19 Sep '19

Hi, This code in dpcm_prune_paths() seems to have no effect: /* is there a valid CODEC DAI widget for this BE */ for_each_rtd_codec_dai(dpcm->be, i, dai) { widget = dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ if (widget && widget_in_list(list, widget)) continue; } Neither of the variables, that are modified in this loop are used outside of it. Was the "continue" maybe meant to continue the encompassing loop? Thanks Guennadi

2 1

[alsa-devel] [PATCH v6 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 19 Sep '19

19 Sep '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v5: - updated slave bindings with proper licencing and fixed up examples. Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 62 + .../soundwire/soundwire-controller.yaml | 82 ++ drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 52 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1345 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

3 9

Re: [alsa-devel] [PATCH V2 3/4] ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_set_runtime_hwparams
by S.j. Wang 19 Sep '19

19 Sep '19

Hi > > When set the runtime hardware parameters, we may need to query the > capability of DMA to complete the parameters. > > This patch is to Extract this operation from > dmaengine_pcm_set_runtime_hwparams function to a separate function > snd_dmaengine_pcm_set_runtime_hwparams, that other components > which need this feature can call this function. > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > --- kbuild test robot report compile issue, will resend v3 after fixing. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ASoC: core: delete component->card_list in soc_remove_component only
by Bard liao 19 Sep '19

19 Sep '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> We add component->card_list in the end of soc_probe_component(). In other words, component->card_list will not be added if there is an error in the soc_probe_component() function. So we can't delete component->card_list in the error handling of soc_probe_component(). Fixes: 22d1423187e5 ("ASoC: soc-core: add soc_cleanup_component()") Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 35f48e9c5ead..aff4b4bf4d07 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -978,7 +978,6 @@ static void soc_cleanup_component(struct snd_soc_component *component) /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); - list_del(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; @@ -991,7 +990,7 @@ static void soc_remove_component(struct snd_soc_component *component) return; snd_soc_component_remove(component); - + list_del(&component->card_list); soc_cleanup_component(component); } -- 2.17.1

5 12

[alsa-devel] [RFC PATCH 0/9] soundwire: add Master device support, GreyBus style
by Pierre-Louis Bossart 18 Sep '19

18 Sep '19

The current use of platform devices leads to limitations in terms of API and problematic reference counts flagged during the review of sysfs patches. As suggested by Greg KH, this patch series introduces a 'Master device' and gets rid of platform devices. The code is inspired by GreyBus, e.g. 'gb_hd_driver' is translated for the SoundWire context as 'sdw_md_driver' and likewise 'gb_host_device' is translated as 'sdw_master_device'. There are differences in the way devices are created, using device_register instead of the multiple steps used in GreyBus. All I know about GreyBus is that it's based on Unipro, so the code updates are based on a first-order analysis, it could very well be I missed concepts while trying to reverse-engineer the code in staging/greybus/. I focused on the concepts mainly and it's also possible that the dual-license for new files or plan-vanilla EXPORT_SYMBOL are not appropriate for all the changes in this series. Please don't take errors as deliberate intent to work-around GPL but rather as an indicator of maturity of the code only developed in the last few days. These changes make it possible to provide new callbacks, e.g splitting the bus startup from the initializations in probe, which is very much desired on Intel platforms to detect the machine driver as early as possible before the hardware is fully powered/enable. These patches will be provided as a separate RFC later today to illustrate the benefit of these changes. To make the code more consistent the first series of patches are renames. I did not rename 'sdw_slave' or 'module_sdw_driver' to avoid compatibility issues since there are codec drivers almost ready for integration. I don't have any specific opinion on if/when additional renames should be done, as long as there is a means to clearly identify what is specific to a SoundWire Master I am fine. Feedback and reviews welcome. Bard Liao (1): soundwire: add device driver to sdw_md_driver Pierre-Louis Bossart (8): soundwire: renames to prepare support for master drivers/devices soundwire: rename dev_to_sdw_dev macro soundwire: rename drv_to_sdw_slave_driver macro soundwire: bus_type: rename sdw_drv_ to sdw_slave_drv soundwire: intel: rename res field as link_res soundwire: add support for sdw_slave_type soundwire: add initial definitions for sdw_master_device soundwire: intel: remove platform devices and provide new interface drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus_type.c | 60 +++--- drivers/soundwire/intel.c | 117 ++++++----- drivers/soundwire/intel.h | 22 +-- drivers/soundwire/intel_init.c | 293 +++++++++++++++++++--------- drivers/soundwire/master.c | 80 ++++++++ drivers/soundwire/slave.c | 9 +- include/linux/soundwire/sdw.h | 39 +++- include/linux/soundwire/sdw_intel.h | 86 +++++++- include/linux/soundwire/sdw_type.h | 34 +++- 12 files changed, 551 insertions(+), 197 deletions(-) create mode 100644 drivers/soundwire/master.c -- 2.20.1

2 17

[alsa-devel] [PATCH] ASoC: rsnd: do error check after rsnd_channel_normalization()
by Kuninori Morimoto 18 Sep '19

18 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSI need to use rsnd_channel_normalization() for TDM-split mode, thus, channel check need to do after that. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark This is for v5.4 sound/soc/sh/rcar/ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f6a7466..fc5d089 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,6 +286,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; + if (rsnd_runtime_is_tdm_split(io)) + chan = rsnd_io_converted_chan(io); + + chan = rsnd_channel_normalization(chan); + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); @@ -300,11 +305,6 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, return 0; } - if (rsnd_runtime_is_tdm_split(io)) - chan = rsnd_io_converted_chan(io); - - chan = rsnd_channel_normalization(chan); - main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- 2.7.4

2 1

[alsa-devel] [PATCH V2 0/4] update supported sample format
by Shengjiu Wang 18 Sep '19

18 Sep '19

This patch serial is to update the supported format for fsl_asrc and fix some format issue. Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_set_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 48 +++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 62 ++------------------ 6 files changed, 181 insertions(+), 89 deletions(-) -- 2.21.0

2 5

[alsa-devel] RFC: TDM mode support on the tegra30
by Ben Dooks 18 Sep '19

18 Sep '19

This is an initial set that adds support for both the TDM and for formats other than 8/16bit audio through the audio system. It also fixes a minor issue with the FIFO stopping order. I have further patches following on which deal with selection of clocks to allow the use of the tegra's audio unit to run as clock slave (all current examples have the tegra as clock master).

6 33

[alsa-devel] Licence change for your alsa-lib UCM commits (Marcin Juszkiewicz: 5 total)
by Jaroslav Kysela 18 Sep '19

18 Sep '19

As discussed, we would like to move the UCM configuration files from the alsa-lib repository to new alsa-ucm-conf repository with the licence change from LGPL-2.1 to BSD-3-Clause. I would like to check, if you agree with this licence change. Please, answer to this e-mail and write your agreement / disagreement (keep CC to the alsa-devel mailing list for the archiving purposes). Thank you for your time and co-operation. Reference: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/153257.html List of your commit(s): f07c14b08280fb461b7f070d55a625e1f5e88a1e conf/ucm: DAISY-I2S: add Samsung ARM Chromebook UCM config from ChromeOS 5b019ea310e6a5d6ff8933bcbca1f3f3471be33b conf/ucm: PandaBoard: add UCM config from Ubuntu 9d0f93af0b35bfdcbd3d758d98715508b69d6008 conf/ucm: PandaBoardES: add UCM config from Ubuntu 7db247be11381bef6e38a41f1d51fa78a99350b5 conf/ucm: SDP4430: add UCM config from Ubuntu 52de910868f56ee7e6b07a1a2ad6a54913f745d0 conf/ucm: tegraalc5632: add UCM config from Ubuntu --- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

15 19

[alsa-devel] 5.1 to stereo using libasound
by Eric Desjardins 18 Sep '19

18 Sep '19

Hi, I am playing a mkv with 5.1 channels from a mkv. I configure libasound for 6 channels and I feed them interleave audio using snd_pcm_writei. I use the default pulse audio device. So far, nothing is wrong. However, I do not ear things correctly. I listen this on my stereo speaker. There is no mixdown to stereo and I was wondering how to do that. I tried to use chmap api but anything I try on the default device returns an error. I was wondering what was the correct way to do that? Using the mixing api? How do we detect that my device output stereo if chmap doesn't work? I am on a ubuntu 18.04 linux box. Thanks, Eric

2 3

[alsa-devel] [PATCH v6 0/4] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 18 Sep '19

18 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. The dependent change on hdmi-codec.c https://patchwork.kernel.org/patch/11047447 has been picked up by Mark Brown in ASoC tree for-5.4 branch. Changes from v5 to v6: 1. Remove the patch for sound/soc/codecs/hdmi-codec.c because it is accepted. 2. Rebase the rest of patches based on drm-misc-next tree. Cheng-Yi Chiang (4): drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 13 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++++- include/drm/bridge/dw_hdmi.h | 4 + sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 6 files changed, 153 insertions(+), 27 deletions(-) -- 2.23.0.237.gc6a4ce50a0-goog

2 6

[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Remove wrong spinlock usage
by Gregory CLEMENT 18 Sep '19

18 Sep '19

A potential bug was reported in the email "[BUG] atmel_ssc_dai: a possible sleep-in-atomic bug in atmel_ssc_shutdown"[1] Indeed in the function atmel_ssc_shutdown() free_irq() was called in a critical section protected by spinlock. However this spinlock is only used in atmel_ssc_shutdown() and atmel_ssc_startup() functions. After further analysis, it occurred that the call to these function are already protected by mutex used on the calling functions. Then we can remove the spinlock which will fix this bug as a side effect. Thanks to this patch the following message disappears: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:909" [1]: https://www.spinics.net/lists/alsa-devel/msg71286.html Reviewed-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Signed-off-by: Gregory CLEMENT <gregory.clement(a)bootlin.com> --- Changelog: v1 -> v2: - Removed the spinlock from the atmel_ssc_info struct - Added the Reviewed-by flag form Alex sound/soc/atmel/atmel_ssc_dai.c | 12 ++---------- sound/soc/atmel/atmel_ssc_dai.h | 1 - 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6f89483ac88c..786b48ae4905 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, @@ -317,13 +314,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, dma_params); - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) return -EBUSY; - } + ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); return 0; } @@ -355,7 +349,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dir_mask = 1 << dir; - spin_lock_irq(&ssc_p->lock); ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { if (ssc_p->initialized) { @@ -369,7 +362,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; ssc_p->forced_divider = 0; } - spin_unlock_irq(&ssc_p->lock); /* Shutdown the SSC clock. */ pr_debug("atmel_ssc_dai: Stopping clock\n"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index ae764cb541c7..3470b966e449 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -93,7 +93,6 @@ struct atmel_ssc_state { struct atmel_ssc_info { char *name; struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ unsigned short initialized; /* true if SSC has been initialized */ unsigned short daifmt; -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: Remove wrong spinlock usage
by Gregory CLEMENT 18 Sep '19

18 Sep '19

A potential bug was reported in the email "[BUG] atmel_ssc_dai: a possible sleep-in-atomic bug in atmel_ssc_shutdown"[1] Indeed in the function atmel_ssc_shutdown() free_irq() was called in a critical section protected by spinlock. However this spinlock is only used in atmel_ssc_shutdown() and atmel_ssc_startup() functions. After further analysis, it occurred that the call to these function are already protected by mutex used on the calling functions. Then we can remove the spinlock which will fix this bug as a side effect. Thanks to this patch the following message disappears: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:909" [1]: https://www.spinics.net/lists/alsa-devel/msg71286.html Signed-off-by: Gregory CLEMENT <gregory.clement(a)bootlin.com> --- sound/soc/atmel/atmel_ssc_dai.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6f89483ac88c..365957e86419 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, @@ -317,13 +314,11 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, dma_params); - spin_lock_irq(&ssc_p->lock); if (ssc_p->dir_mask & dir_mask) { spin_unlock_irq(&ssc_p->lock); return -EBUSY; } ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); return 0; } @@ -355,7 +350,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dir_mask = 1 << dir; - spin_lock_irq(&ssc_p->lock); ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { if (ssc_p->initialized) { @@ -369,7 +363,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; ssc_p->forced_divider = 0; } - spin_unlock_irq(&ssc_p->lock); /* Shutdown the SSC clock. */ pr_debug("atmel_ssc_dai: Stopping clock\n"); -- 2.23.0

2 2

[alsa-devel] alsa acting up
by liebrecht＠grossmann-venter.com 18 Sep '19

18 Sep '19

See images in links. 1) "http://grossmann-venter.com/visitors/ug/alsa/alsa-pulse-181vsl_unplug.png" 2) "http://grossmann-venter.com/visitors/ug/alsa/alsa-dont recognize1818VSLpng" 1) As soon as pulseaudio is started Alsa says 1818VSL is unplugged....? 2) Clearly Presonus 1818VSL is found as lsusb output says, but alsa has no clue about it. How do I get alsa to recognise 181VSL as it is found by lsusb so it is presnet on the system. It used to work fine but alsa just stopped working with 1818VSL recently for no reason. I know this is a developers group, but please at least answer where I can get help. Alsa help is absolutely non-existing. I also posted a bug with no response in another thread. Thanks.

1 0

[alsa-devel] [PATCH v5 0/5] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 18 Sep '19

18 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. Changes from v4 to v5: - synopsys/Kconfig: Remove the incorrect dependency change in v4. - rockchip/Kconfig: Add dependency of hdmi-codec when it is really need for jack support. Cheng-Yi Chiang (5): ASoC: hdmi-codec: Add an op to set callback function for plug event drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 13 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++++- include/drm/bridge/dw_hdmi.h | 4 + include/sound/hdmi-codec.h | 17 +++ sound/soc/codecs/hdmi-codec.c | 46 +++++++ sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 8 files changed, 216 insertions(+), 27 deletions(-) -- 2.22.0.510.g264f2c817a-goog

4 12

[alsa-devel] alsa acting up
by liebrecht＠grossmann-venter.com 18 Sep '19

18 Sep '19

See images in links. 1) "http://grossmann-venter.com/visitors/ug/alsa/alsa-pulse-181vsl_unplug.png" 2) "http://grossmann-venter.com/visitors/ug/alsa/alsa-dont recognize1818VSLpng" 1) As soon as pulseaudio is started Alsa says 1818VSL is unplugged....? 2) Clearly Presonus 1818VSL is found as lsusb output says, but alsa has no clue about it. How do I get alsa to recognise 181VSL as it is found by lsusb so it is presnet on the system. It used to work fine but alsa just stopped working with 1818VSL recently for no reason. I know this is a developers group, but please at least answer where I can get help. Alsa help is absolutely non-existing. I also posted a bug with no response in another thread. Thanks.

1 0

Re: [alsa-devel] Alsa-devel Digest, Vol 149, Issue 28
by Channaiah Vanitha (RBEI/ECF3) 18 Sep '19

18 Sep '19

Hello Jaroslav-san, This mail is regarding : 3. Re: GIT: Regarding the issue we are facing in the commit 37728639ae05de702825d96bd1d42e24ae772248 (Jaroslav Kysela) Dne 03. 07. 19 v 14:17 Channaiah Vanitha (RBEI/ECF3) napsal(a): > Hello Jaroslav-san, > >> I think that it would be probably best to force the parameters for your hardware (--period-size and --buffer-size arguments for aplay or the time counterparts - --period-time and --buffer-time). The refining rules might not select the perfect configuration in some cases. > > I tried to force parameters "period-size" in multiples of 2ms as our hardware supports 2ms period time data and buffer-size=twice period size. > But still I face the issue. There is no exact 2ms period size for the rate 11025Hz, because it's float number 21.9780 (period size)... You may try values like 15,35,45,105 (anything which can match 11025 / PERIOD_SIZE without the remainder). Jaroslav We tried executing the playback use-case with below parameters as suggested. But still we able to see: Unable to install hw params. Kindly provide your feedback. The output from below Hardware : a. RCAR salvator-xs which supports 2 channel. b. IMX which supports 8 channel. root@rcar-gen3:~# aplay -v -Dentertainment_main -r11025 -c6 -fS16_LE --period-size=15 /dev/random Playing raw data '/dev/random' : Signed 16 bit Little Endian, Rate 11025 Hz, Channels 6 aplay: set_params:1411: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 96 CHANNELS: 6 RATE: NONE PERIOD_TIME: 2000 PERIOD_SIZE: NONE PERIOD_BYTES: (264 276) PERIODS: (15 16) BUFFER_TIME: (31927 31928) BUFFER_SIZE: 352 BUFFER_BYTES: 4224 TICK_TIME: 0 root@rcar-gen3:~# aplay -v -Dentertainment_main -r11025 -c6 -fS16_LE --period-size=105 /dev/random Playing raw data '/dev/random' : Signed 16 bit Little Endian, Rate 11025 Hz, Channels 6 aplay: set_params:1411: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 96 CHANNELS: 6 RATE: NONE PERIOD_TIME: 10000 PERIOD_SIZE: NONE PERIOD_BYTES: (1320 1332) PERIODS: (2 3) BUFFER_TIME: (29931 29932) BUFFER_SIZE: 330 BUFFER_BYTES: 3960 TICK_TIME: 0 root@rcar-gen3:~# aplay -v -Dentertainment_main -r11025 -c6 -fS16_LE --period-size=35 /dev/random Playing raw data '/dev/random' : Signed 16 bit Little Endian, Rate 11025 Hz, Channels 6 aplay: set_params:1411: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 96 CHANNELS: 6 RATE: NONE PERIOD_TIME: 4000 PERIOD_SIZE: NONE PERIOD_BYTES: (528 540) PERIODS: (7 8) BUFFER_TIME: (31927 31928) BUFFER_SIZE: 352 BUFFER_BYTES: 4224 TICK_TIME: 0 root@rcar-gen3:~# aplay -v -Dentertainment_main -r11025 -c6 -fS16_LE --period-size=45 /dev/random Playing raw data '/dev/random' : Signed 16 bit Little Endian, Rate 11025 Hz, Channels 6 aplay: set_params:1411: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 96 CHANNELS: 6 RATE: NONE PERIOD_TIME: 4000 PERIOD_SIZE: NONE PERIOD_BYTES: (528 540) PERIODS: (7 8) BUFFER_TIME: (31927 31928) BUFFER_SIZE: 352 BUFFER_BYTES: 4224 TICK_TIME: 0 Best regards, Vanitha Channaiah RBEI/ECF3 Tel. +91 80 6136-4436 -----Original Message----- From: Alsa-devel <alsa-devel-bounces(a)alsa-project.org> On Behalf Of alsa-devel-request(a)alsa-project.org Sent: Thursday, July 4, 2019 5:32 PM To: alsa-devel(a)alsa-project.org Subject: Alsa-devel Digest, Vol 149, Issue 28 Send Alsa-devel mailing list submissions to alsa-devel(a)alsa-project.org To subscribe or unsubscribe via the World Wide Web, visit https://mailman.alsa-project.org/mailman/listinfo/alsa-devel or, via email, send a message with subject or body 'help' to alsa-devel-request(a)alsa-project.org You can reach the person managing the list at alsa-devel-owner(a)alsa-project.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Alsa-devel digest..." Today's Topics: 1. Re: [PATCH v2 1/2] ASoC: SOF: add flag for position update ipc (Kai Vehmanen) 2. Re: [PATCH v2 2/2] ASoC: SOF: Intel: fix reset of host_period_bytes (Kai Vehmanen) 3. Re: GIT: Regarding the issue we are facing in the commit 37728639ae05de702825d96bd1d42e24ae772248 (Jaroslav Kysela) 4. Re: [PATCH v2 1/2] ASoC: SOF: add flag for position update ipc (Keyon Jie) 5. bug in playing 6-channel sound (Julian Bradfield) 6. Re: [PATCH v2 1/2] ASoC: SOF: add flag for position update ipc (Kai Vehmanen) 7. Re: [PATCH v2 34/35] sound/soc/codecs: Use kmemdup rather than duplicating its implementation (Richard Fitzgerald) 8. audio lost from speaker after reboot from windows on the device ALC295 (He, Bo) ---------------------------------------------------------------------- Message: 1 Date: Thu, 4 Jul 2019 13:03:06 +0300 (EEST) From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> To: Keyon Jie <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org, ranjani.sridharan(a)linux.intel.com, Marcin Rajwa <marcin.rajwa(a)linux.intel.com>, pierre-louis.bossart(a)linux.intel.com Subject: Re: [alsa-devel] [PATCH v2 1/2] ASoC: SOF: add flag for position update ipc Message-ID: <alpine.DEB.2.21.1907041254100.4409@eliteleevi> Content-Type: text/plain; charset=US-ASCII Hi, patch looks good, but commit message could be improved. On Wed, 3 Jul 2019, Keyon Jie wrote: > In some cases, FW might need use the host_period_bytes even no > position update ipc reqiured from driver, here add another flag for > position update, and preserve host_period_bytes for FW to use. Speculation on what FW might do is not really needed. The host_period_bytes field has been overloaded with multiple semantics and this patch clears that, right. How about: "" Remove the special case semantics of 'host_period_bytes==0'. Add a new field 'no_period_irq' to signal whether period elapsed IPC should be sent and use 'host_period_bytes' only for period size. "" > This might require corresponding FW change and ABI alignment. This is not helpful -- we know this _is_ a minor ABI change and needs to be aligned with FW. Br, Kai ------------------------------ Message: 2 Date: Thu, 4 Jul 2019 13:05:21 +0300 (EEST) From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> To: Keyon Jie <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org, ranjani.sridharan(a)linux.intel.com, Marcin Rajwa <marcin.rajwa(a)linux.intel.com>, pierre-louis.bossart(a)linux.intel.com Subject: Re: [alsa-devel] [PATCH v2 2/2] ASoC: SOF: Intel: fix reset of host_period_bytes Message-ID: <alpine.DEB.2.21.1907041304101.4409@eliteleevi> Content-Type: text/plain; charset=US-ASCII Hi, On Wed, 3 Jul 2019, Keyon Jie wrote: > From: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> > > This patch prevents the reset of host period bytes. > The parameter has been used to keep information about completion of > period copy. Right now we keep this information in period_irq. > > Signed-off-by: Marcin Rajwa <marcin.rajwa(a)linux.intel.com> > Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> looks good, for this patch: Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> ------------------------------ Message: 3 Date: Thu, 4 Jul 2019 12:18:24 +0200 From: Jaroslav Kysela <perex(a)perex.cz> To: "Channaiah Vanitha (RBEI/ECF3)" <Vanitha.Channaiah(a)in.bosch.com> Cc: "Wischer Timo $ADITG/ESS$" <twischer(a)de.adit-jv.com>, "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> Subject: Re: [alsa-devel] GIT: Regarding the issue we are facing in the commit 37728639ae05de702825d96bd1d42e24ae772248 Message-ID: <556614e1-acba-635c-beed-6484fb887299(a)perex.cz> Content-Type: text/plain; charset=utf-8 Dne 03. 07. 19 v 14:17 Channaiah Vanitha (RBEI/ECF3) napsal(a): > Hello Jaroslav-san, > >> I think that it would be probably best to force the parameters for your hardware (--period-size and --buffer-size arguments for aplay or the time counterparts - --period-time and --buffer-time). The refining rules might not select the perfect configuration in some cases. > > I tried to force parameters "period-size" in multiples of 2ms as our hardware supports 2ms period time data and buffer-size=twice period size. > But still I face the issue. There is no exact 2ms period size for the rate 11025Hz, because it's float number 21.9780 (period size)... You may try values like 15,35,45,105 (anything which can match 11025 / PERIOD_SIZE without the remainder). Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc. ------------------------------ Message: 4 Date: Thu, 4 Jul 2019 18:29:20 +0800 From: Keyon Jie <yang.jie(a)linux.intel.com> To: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org, ranjani.sridharan(a)linux.intel.com, Marcin Rajwa <marcin.rajwa(a)linux.intel.com>, pierre-louis.bossart(a)linux.intel.com Subject: Re: [alsa-devel] [PATCH v2 1/2] ASoC: SOF: add flag for position update ipc Message-ID: <3788800e-7050-d68a-bec6-5b443fd0d563(a)linux.intel.com> Content-Type: text/plain; charset=utf-8; format=flowed On 2019/7/4 ??6:03, Kai Vehmanen wrote: > Hi, > > patch looks good, but commit message could be improved. > > On Wed, 3 Jul 2019, Keyon Jie wrote: > >> In some cases, FW might need use the host_period_bytes even no position >> update ipc reqiured from driver, here add another flag for position update, >> and preserve host_period_bytes for FW to use. > > Speculation on what FW might do is not really needed. The > host_period_bytes field has been overloaded with multiple > semantics and this patch clears that, right. How about: Well, to me it is flavor choice, Ranjani suggested to illustrate the use case where the FW will use this host_period_bytes, and I agreed this will help people to understand why we need this change. > > "" > Remove the special case semantics of 'host_period_bytes==0'. > Add a new field 'no_period_irq' to signal whether period elapsed > IPC should be sent and use 'host_period_bytes' only for > period size. > "" > >> This might require corresponding FW change and ABI alignment. > > This is not helpful -- we know this _is_ a minor ABI change > and needs to be aligned with FW. It is minor change, but the FW change is still required, otherwise, we will get extra position update IPCs which may confuse the driver, please refer to here: https://github.com/thesofproject/sof/pull/1592 Thanks, ~Keyon > > Br, Kai > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > https://mailman.alsa-project.org/mailman/listinfo/alsa-devel > ------------------------------ Message: 5 Date: Thu, 4 Jul 2019 11:46:20 +0100 From: Julian Bradfield <jalsa-devel(a)jcbradfield.org> To: alsa-devel(a)alsa-project.org Subject: [alsa-devel] bug in playing 6-channel sound Message-ID: <23837.55548.631053.741245(a)vis.stevens-bradfield.com> Content-Type: text/plain; charset=us-ascii Some five years ago, the following post https://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/072265.… reported a failure to play six-channel sound, that could be worked around by setting the prealloc value for the device to various values *other than* the default 64. I have just run into this playing a movie from my laptop (Thinkpad Yoga X1, Intel HDA PCH sound card outputting through HDMI) running Xubuntu 18.04, up to date as at the time of posting. I see exactly the same problem as in that post, and the same workaround works for me - I tried prealloc values 63,65,32,128, and any of them is ok. I'm a bit baffled that such a problem has survived for several years. If any developer would like me to provide debugging information, please get in touch - I have some time available at present, but not enough to learn ALSA and debug it myself! I will remain subscribed to the list for a few weeks to read any followups. ------------------------------ Message: 6 Date: Thu, 4 Jul 2019 14:00:31 +0300 (EEST) From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> To: Keyon Jie <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org, Marcin Rajwa <marcin.rajwa(a)linux.intel.com>, ranjani.sridharan(a)linux.intel.com, Kai Vehmanen <kai.vehmanen(a)linux.intel.com>, pierre-louis.bossart(a)linux.intel.com Subject: Re: [alsa-devel] [PATCH v2 1/2] ASoC: SOF: add flag for position update ipc Message-ID: <alpine.DEB.2.21.1907041344240.4409@eliteleevi> Content-Type: text/plain; charset=US-ASCII Hi, On Thu, 4 Jul 2019, Keyon Jie wrote: > Well, to me it is flavor choice, Ranjani suggested to illustrate the use > case where the FW will use this host_period_bytes, and I agreed this > will help people to understand why we need this change. hmm, ok. So maybe add "Allows FW to use 'host_period_bytes' field for its original purpose" to my proposed wording..? >> This is not helpful -- we know this _is_ a minor ABI change >> and needs to be aligned with FW. > It is minor change, but the FW change is still required, otherwise, we > will get extra position update IPCs which may confuse the driver, please [...] > https://github.com/thesofproject/sof/pull/1592 Ack, but we know this already so best to put the accurate description in the commit message. The "might require FW change" is a bit scary statement in a patch touching ABI structs. ;) ------------------------------ Message: 7 Date: Thu, 4 Jul 2019 12:47:29 +0100 From: Richard Fitzgerald <rf(a)opensource.cirrus.com> To: Fuqian Huang <huangfq.daxian(a)gmail.com> Cc: alsa-devel(a)alsa-project.org, patches(a)opensource.cirrus.com, Takashi Iwai <tiwai(a)suse.com>, Liam Girdwood <lgirdwood(a)gmail.com>, Mark Brown <broonie(a)kernel.org>, linux-kernel(a)vger.kernel.org Subject: Re: [alsa-devel] [PATCH v2 34/35] sound/soc/codecs: Use kmemdup rather than duplicating its implementation Message-ID: <52ee7351-19fc-4fd3-33f8-9392a4ad9526(a)opensource.cirrus.com> Content-Type: text/plain; charset="utf-8"; format=flowed Commit message title prefix should be "ASoC: wm0010:" not "sound/soc /codecs:". Take a look at other patches to the same files. > kmemdup is introduced to duplicate a region of memory in a neat way. > Rather than kmalloc/kzalloc + memcpy, which the programmer needs to > write the size twice (sometimes lead to mistakes), kmemdup improves > readability, leads to smaller code and also reduce the chances of mistakes. > Suggestion to use kmemdup rather than using kmalloc/kzalloc + memcpy. > > Acked-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> > Signed-off-by: Fuqian Huang <huangfq.daxian(a)gmail.com> > --- > Changes in v2: > - Fix a typo in commit message (memset -> memcpy) > - Split into two patches > > sound/soc/codecs/wm0010.c | 4 +--- > 1 file changed, 1 insertion(+), 3 deletions(-) > > diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c > index 727d6703c905..807826f30f58 100644 > --- a/sound/soc/codecs/wm0010.c > +++ b/sound/soc/codecs/wm0010.c > @@ -515,7 +515,7 @@ static int wm0010_stage2_load(struct snd_soc_component *component) > dev_dbg(component->dev, "Downloading %zu byte stage 2 loader\n", fw->size); > > /* Copy to local buffer first as vmalloc causes problems for dma */ > - img = kzalloc(fw->size, GFP_KERNEL | GFP_DMA); > + img = kmemdup(&fw->data[0], fw->size, GFP_KERNEL | GFP_DMA); > if (!img) { > ret = -ENOMEM; > goto abort2; > @@ -527,8 +527,6 @@ static int wm0010_stage2_load(struct snd_soc_component *component) > goto abort1; > } > > - memcpy(img, &fw->data[0], fw->size); > - > spi_message_init(&m); > memset(&t, 0, sizeof(t)); > t.rx_buf = out; > ------------------------------ Message: 8 Date: Thu, 4 Jul 2019 12:02:17 +0000 From: "He, Bo" <bo.he(a)intel.com> To: "kailang(a)realtek.com" <kailang(a)realtek.com>, "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org>, "linux-kernel(a)vger.kernel.org" <linux-kernel(a)vger.kernel.org> Cc: "chiu(a)endlessm.com" <chiu(a)endlessm.com>, "tiwai(a)suse.com" <tiwai(a)suse.com>, "drake(a)endlessm.com" <drake(a)endlessm.com>, "hui.wang(a)canonical.com" <hui.wang(a)canonical.com>, "jian-hong(a)endlessm.com" <jian-hong(a)endlessm.com> Subject: [alsa-devel] audio lost from speaker after reboot from windows on the device ALC295 Message-ID: <CD6925E8781EFD4D8E11882D20FC406D52AB58B6(a)SHSMSX104.ccr.corp.intel.com> Content-Type: text/plain; charset="us-ascii" Hi, patch_realtek.c maintainer: I see one issue that reboot from windows and boot to ubuntu, the audio lost from speaker, I suspect there are some bugs in patch_realtek.c drivers, the device is ALC295 and the device id is 0x10ec0295. I have done the below experiments: 1. reboot from windows to windows, the audio is persist . 2. reboot from windows to ubuntu, the audio lost from speaker, but can hear if I hotplug one earphone. 3. if the issue reproduce after reboot from windows, reboot the ubuntu can't restore the audio, I suspect it's warm reset. 4. if I write the port 0xcf9 with 0xe to do cold reset, the audio can restore. 5. if I do suspend/resume, the audio can restore, I suspect do cold boot and suspend will trigger the platform reset to reset the ALC295. 6. if I do double function reset (write the verb 0x7ff in alc_init), the audio is still can't restore. snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_CODEC_RESET, 0); /* Function reset */ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_CODEC_RESET, 0); /* double Function reset */ 7. the issue is first found on kernel 4.19.50, I still see the issue with the latest kernel 5.2-rc2, is it possible windows change some default registers, but ALC295 don't initialize the register? ------------------------------ Subject: Digest Footer _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org https://mailman.alsa-project.org/mailman/listinfo/alsa-devel ------------------------------ End of Alsa-devel Digest, Vol 149, Issue 28 *******************************************

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Re: [alsa-devel] [PATCH 2/3] ASoC: fsl_asrc: update supported sample format
by S.j. Wang 18 Sep '19

18 Sep '19

Hi > > On Fri, Sep 13, 2019 at 05:48:40AM +0000, S.j. Wang wrote: > > Hi > > > > > > > > On Tue, Sep 10, 2019 at 02:07:25AM +0000, S.j. Wang wrote: > > > > > On Mon, Sep 09, 2019 at 06:33:20PM -0400, Shengjiu Wang wrote: > > > > > > The ASRC support 24bit/16bit/8bit input width, so S20_3LE > > > > > > format should not be supported, it is word width is 20bit. > > > > > > > > > > I thought 3LE used 24-bit physical width. And the driver assigns > > > > > ASRC_WIDTH_24_BIT to "width" for all non-16bit cases, so 20-bit > > > > > would go for that 24-bit slot also. I don't clearly recall if I > > > > > had explicitly tested S20_3LE, but I feel it should work since I put > there... > > > > > > > > For S20_3LE, the width is 20bit, but the ASRC only support 24bit, > > > > if set the ASRMCR1n.IWD= 24bit, because the actual width is 20 > > > > bit, the volume is Lower than expected, it likes 24bit data right shift 4 > bit. > > > > So it is not supported. > > > > > > Hmm..S20_3LE right-aligns 20 bits in a 24-bit slot? I thought > > > they're left aligned... > > > > > > If this is the case...shouldn't we have the same lower-volume > > > problem for all hardwares that support S20_3LE now? > > > > Actually some hardware/module when they do transmission from FIFO to > > shift register, they can select the start bit, for example from the > > 20th bit. but not all module have this capability. > > > > For ASRC, it haven't. IWD can only cover the data width, there is no > > Other bit for slot width. > > Okay..let's drop the S20_3LE then. But would it be possible for you to > elaborate the reasoning into the commit message also? Just for case when > people ask why we remove it simply. > > Thanks OK. Best regards Wang shengjiu

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[alsa-devel] Build failure after merge of the sound-current
by Mark Brown 18 Sep '19

18 Sep '19

After merging the sound-current tree, today's linux-next build for x86 allmodconfig failed like this: In file included from /home/broonie/next/next/sound/soc/intel/boards/sof_rt5682.c:23: /home/broonie/next/next/sound/soc/intel/boards/../common/soc-intel-quirks.h: In function 'soc_intel_is_cml': /home/broonie/next/next/sound/soc/intel/boards/../common/soc-intel-quirks.h:39:23: error: 'INTEL_FAM6_KABYLAKE_MOBILE' undeclared (first use in this function); did you mean 'INTEL_FAM6_KABYLAKE_L'? SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); ^~~~~~~~~~~~~~~~~~~~~~~~~~ /home/broonie/next/next/sound/soc/intel/boards/../common/soc-intel-quirks.h:18:44: note: in definition of macro 'ICPU' #define ICPU(model) { X86_VENDOR_INTEL, 6, model, X86_FEATURE_ANY, } ^~~~~ /home/broonie/next/next/sound/soc/intel/boards/../common/soc-intel-quirks.h:39:1: note: in expansion of macro 'SOC_INTEL_IS_CPU' SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); ^~~~~~~~~~~~~~~~ /home/broonie/next/next/sound/soc/intel/boards/../common/soc-intel-quirks.h:39:23: note: each undeclared identifier is reported only once for each function it appears in SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); ^~~~~~~~~~~~~~~~~~~~~~~~~~ /home/broonie/next/next/sound/soc/intel/boards/../common/soc-intel-quirks.h:18:44: note: in definition of macro 'ICPU' #define ICPU(model) { X86_VENDOR_INTEL, 6, model, X86_FEATURE_ANY, } ^~~~~ /home/broonie/next/next/sound/soc/intel/boards/../common/soc-intel-quirks.h:39:1: note: in expansion of macro 'SOC_INTEL_IS_CPU' SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); ^~~~~~~~~~~~~~~~ make[5]: *** [/home/broonie/next/next/scripts/Makefile.build:281: sound/soc/intel/boards/sof_rt5682.o] Error 1 make[4]: *** [/home/broonie/next/next/scripts/Makefile.build:497: sound/soc/intel/boards] Error 2 make[4]: *** Waiting for unfinished jobs.... In file included from /home/broonie/next/next/sound/soc/intel/atom/sst/sst_acpi.c:35: /home/broonie/next/next/sound/soc/intel/atom/sst/../../common/soc-intel-quirks.h: In function 'soc_intel_is_cml': /home/broonie/next/next/sound/soc/intel/atom/sst/../../common/soc-intel-quirks.h:39:23: error: 'INTEL_FAM6_KABYLAKE_MOBILE' undeclared (first use in this function); did you mean 'INTEL_FAM6_KABYLAKE_L'? SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); ^~~~~~~~~~~~~~~~~~~~~~~~~~ /home/broonie/next/next/sound/soc/intel/atom/sst/../../common/soc-intel-quirks.h:18:44: note: in definition of macro 'ICPU' #define ICPU(model) { X86_VENDOR_INTEL, 6, model, X86_FEATURE_ANY, } ^~~~~ /home/broonie/next/next/sound/soc/intel/atom/sst/../../common/soc-intel-quirks.h:39:1: note: in expansion of macro 'SOC_INTEL_IS_CPU' SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); ^~~~~~~~~~~~~~~~ /home/broonie/next/next/sound/soc/intel/atom/sst/../../common/soc-intel-quirks.h:39:23: note: each undeclared identifier is reported only once for each function it appears in SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); ^~~~~~~~~~~~~~~~~~~~~~~~~~ /home/broonie/next/next/sound/soc/intel/atom/sst/../../common/soc-intel-quirks.h:18:44: note: in definition of macro 'ICPU' #define ICPU(model) { X86_VENDOR_INTEL, 6, model, X86_FEATURE_ANY, } ^~~~~ /home/broonie/next/next/sound/soc/intel/atom/sst/../../common/soc-intel-quirks.h:39:1: note: in expansion of macro 'SOC_INTEL_IS_CPU' SOC_INTEL_IS_CPU(cml, INTEL_FAM6_KABYLAKE_MOBILE); ^~~~~~~~~~~~~~~~ make[6]: *** [/home/broonie/next/next/scripts/Makefile.build:281: sound/soc/intel/atom/sst/sst_acpi.o] Error 1 make[5]: *** [/home/broonie/next/next/scripts/Makefile.build:497: sound/soc/intel/atom/sst] Error 2 make[4]: *** [/home/broonie/next/next/scripts/Makefile.build:497: sound/soc/intel/atom] Error 2 make[3]: *** [/home/broonie/next/next/scripts/Makefile.build:497: sound/soc/intel] Error 2 make[3]: *** Waiting for unfinished jobs.... make[2]: *** [/home/broonie/next/next/scripts/Makefile.build:497: sound/soc] Error 2 make[1]: *** [/home/broonie/next/next/Makefile:1087: sound] Error 2 make[1]: *** Waiting for unfinished jobs.... make: *** [Makefile:179: sub-make] Error 2 Command exited with non-zero status 2 caused by commit c643c189f0fec116 (ASoC: Intel: boards: Add Cometlake machine driver support) IIRC there was some previous discussion of a dependency on other changes here but I thought that was resolved? I've reverted that commit for today.

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[alsa-devel] [PATCH v2 0/3] Several SAI fixes
by Daniel Baluta 17 Sep '19

17 Sep '19

This series contains several fixes for SAI. They are unrelated but grouped them in a patch series to be easier applied. Daniel Baluta (1): ASoC: fsl_sai: Fix TCSR.TE/RCSR.RE in synchronous mode Mihai Serban (1): ASoC: fsl_sai: Fix noise when using EDMA Shengjiu Wang (1): ASoC: fsl_sai: Fix xMR setting in synchronous mode sound/soc/fsl/fsl_sai.c | 31 ++++++++++++++++++++++--------- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 23 insertions(+), 9 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v4 0/9] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 17 Sep '19

17 Sep '19

Hi all, here's the 4th 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. v4 changes: - Change order of patches to not break bisect (Pierre's feedback). - Improve the explanation in commit message for mst_no_extra_pcms, patch 1. (Pierre's feedback). - Fix errors in PCM constraints for HDMI (Pierre's feedback). - Fix an issue on Ice Lake platforms (patch 3). 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 | 100 +++++++++++++++--- sound/soc/codecs/hdac_hda.h | 12 ++- 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 | 19 +++- sound/soc/sof/intel/hda.c | 6 ++ sound/soc/sof/intel/hda.h | 6 +- 17 files changed, 323 insertions(+), 27 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

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[alsa-devel] [PATCH v5] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 17 Sep '19

17 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

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[alsa-devel] [PATCH] ASoC: rt1011: set tx/rx slots from tx/rx_mask in TDM case
by shumingf＠realtek.com 17 Sep '19

17 Sep '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 | 97 +++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt1011.h | 23 ++++++++++ 2 files changed, 115 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index be1e276e3631..a36144de11cf 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) @@ -1850,6 +1850,91 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } + /* 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_; + } + if (rx_slotnum) { + if (first_bit == 0 || first_bit == 2 || + first_bit == 4 || first_bit == 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)); + } else { + 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)); + } + } + + /* 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); + if (first_bit == 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); + else if (first_bit == 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); + else if (first_bit == 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); + else if (first_bit == 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); + } else if (tx_slotnum == 2) { + if (first_bit == 0 || first_bit == 2 || + first_bit == 4 || first_bit == 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); + } else { + 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, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); @@ -1860,10 +1945,12 @@ 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] ALSA: usb-audio: Add Hiby device family to quirks for native DSD support
by ilya.pshonkin＠netforce.ua 17 Sep '19

17 Sep '19

From: Ilya Pshonkin <sudokamikaze(a)protonmail.com> This patch adds quirk VID ID for Hiby portable players family with native DSD playback support Signed-off-by: Ilya Pshonkin <sudokamikaze(a)protonmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 78858918cbc1..64a8d73972e3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1655,6 +1655,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ case 0x2ab6: /* T+A devices */ + case 0xc502: /* HiBy devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.23.0

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[alsa-devel] [PATCH] ALSA: hda/realtek - Fix alienware headset mic
by James McDonnell 17 Sep '19

17 Sep '19

From: James McDonnell <james_mcdonnell(a)hotmail.com> Subject: [PATCH] ALSA: hda/realtek - Fix alienware headset mic Headset microphone quirk for alienware 15r3. Without this using a headset with mic attached will not work. Signed-off-by: James McDonnell <james_mcdonnell(a)hotmail.com> Cc: alsa-devel(a)alsa-project.org --- sound/pci/hda/patch_realtek.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e333b3e30e31..347fcd755571 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5745,6 +5745,7 @@ enum { ALC292_FIXUP_DELL_E7X, ALC292_FIXUP_DISABLE_AAMIX, ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK, + ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE, ALC298_FIXUP_DELL1_MIC_NO_PRESENCE, ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, @@ -6426,6 +6427,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC292_FIXUP_DISABLE_AAMIX }, + [ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* headset mic w/o jack detect */ + { } + }, + .chained_before = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE, + }, [ALC298_FIXUP_DELL1_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -7680,6 +7690,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x17, 0x90170110}, {0x1a, 0x03011020}, {0x21, 0x03211030}), + SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_ALIENWARE_MIC_NO_PRESENCE, + {0x12, 0xb7a60140}, + {0x17, 0x90170110}, + {0x1a, 0x03a11030}, + {0x21, 0x03211020}), SND_HDA_PIN_QUIRK(0x10ec0299, 0x1028, "Dell", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, ALC225_STANDARD_PINS, {0x12, 0xb7a60130}, -- 2.23.0

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[alsa-devel] [PATCH] ASoC: Intel: cht_bsw_rt5645: Add quirk for boards using pmc_plt_clk_0
by Sam McNally 17 Sep '19

17 Sep '19

As of commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL"), the cht_bsw_rt5645 driver needs to enable the clock it's using for the codec's mclk. It does this from commit 7735bce05a9c ("ASoC: Intel: boards: use devm_clk_get() unconditionally"), enabling pmc_plt_clk_3. However, Strago family Chromebooks use pmc_plt_clk_0 for the codec mclk, resulting in white noise with some digital microphones. Add a DMI-based quirk for Strago family Chromebooks to use pmc_plt_clk_0 instead. Signed-off-by: Sam McNally <sammc(a)chromium.org> --- sound/soc/intel/boards/cht_bsw_rt5645.c | 26 +++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 8879c3be29d5..c68a5b85a4a0 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -48,6 +48,7 @@ struct cht_mc_private { #define CHT_RT5645_SSP2_AIF2 BIT(16) /* default is using AIF1 */ #define CHT_RT5645_SSP0_AIF1 BIT(17) #define CHT_RT5645_SSP0_AIF2 BIT(18) +#define CHT_RT5645_PMC_PLT_CLK_0 BIT(19) static unsigned long cht_rt5645_quirk = 0; @@ -59,6 +60,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled"); if (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + dev_info(dev, "quirk PMC_PLT_CLK_0 enabled"); } static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -226,15 +229,21 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, return 0; } -/* uncomment when we have a real quirk static int cht_rt5645_quirk_cb(const struct dmi_system_id *id) { cht_rt5645_quirk = (unsigned long)id->driver_data; return 1; } -*/ static const struct dmi_system_id cht_rt5645_quirk_table[] = { + { + /* Strago family Chromebooks */ + .callback = cht_rt5645_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_Strago"), + }, + .driver_data = (void *)CHT_RT5645_PMC_PLT_CLK_0, + }, { }, }; @@ -526,6 +535,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int dai_index = 0; int ret_val = 0; int i; + const char *mclk_name; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -662,11 +672,15 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (ret_val) return ret_val; - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + dev_err(&pdev->dev, "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.23.0.237.gc6a4ce50a0-goog

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Re: [alsa-devel] [PATCH 2/3] ASoC: fsl_asrc: update supported sample format
by S.j. Wang 17 Sep '19

17 Sep '19

Hi > > On Mon, Sep 09, 2019 at 06:33:20PM -0400, Shengjiu Wang wrote: > > The ASRC support 24bit/16bit/8bit input width, so S20_3LE format > > should not be supported, it is word width is 20bit. > > I thought 3LE used 24-bit physical width. And the driver assigns > ASRC_WIDTH_24_BIT to "width" for all non-16bit cases, so 20-bit would go > for that 24-bit slot also. I don't clearly recall if I had explicitly tested > S20_3LE, but I feel it should work since I put there... > > Thanks > Nicolin > For S20_3LE, the width is 20bit, but the ASRC only support 24bit, if set the ASRMCR1n.IWD= 24bit, because the actual width is 20 bit, the volume is Lower than expected, it likes 24bit data right shift 4 bit. So it is not supported. Best regards Wang shengjiu

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[alsa-devel] [RFC 00/15] Add Samus Hotwording for RT5677
by Curtis Malainey 17 Sep '19

17 Sep '19

This patch series adds the hotwording implementation used in the Pixelbook on the RT5677 driver. Known Issues: There is a known issue where the system will fail to detect a hotword if suspended while the stream is open. This is due to the fact that the haswell-dsp suspends its I2S MCLK before the RT5677 suspends which causes the writes and reads to become corrupted as a result. Any recommendations to correct this behaviour would be appreciated. Ben Zhang (12): ASoC: rt5677: keep analog power register at SND_SOC_BIAS_OFF ASoC: rt5677: Add a PCM device for streaming hotword via SPI ASoC: rt5677: Load firmware via SPI ASoC: rt5677: Auto enable/disable DSP for hotwording ASoC: bdw-rt5677: Add a DAI link for rt5677 SPI PCM device ASoC: rt5677: Enable jack detect while DSP is running ASoC: rt5677: Use delayed work for DSP firmware load ASoC: rt5677: Add DAPM audio path for hotword stream ASoC: rt5677: Mark reg RT5677_PWR_ANLG2 as volatile ASoC: rt5677: Stop and restart DSP over suspend/resume ASoC: rt5677: Transfer one period at a time over SPI ASoC: rt5677: Disable irq at suspend Curtis Malainey (3): ASoC: rt5677: Remove magic number register writes ASoC: rt5677: Allow VAD to be shut on/off at all times ASoC: rt5677: Turn on MCLK1 for DSP via DAPM sound/soc/codecs/rt5677-spi.c | 401 ++++++++++++++++++++++ sound/soc/codecs/rt5677-spi.h | 1 + sound/soc/codecs/rt5677.c | 496 +++++++++++++++++++++++----- sound/soc/codecs/rt5677.h | 9 +- sound/soc/intel/boards/bdw-rt5677.c | 18 + 5 files changed, 843 insertions(+), 82 deletions(-) -- 2.23.0.187.g17f5b7556c-goog

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[alsa-devel] [PATCH] ALSA: dice: fix wrong packet parameter for Alesis iO26
by Takashi Sakamoto 17 Sep '19

17 Sep '19

At higher sampling rate (e.g. 192.0 kHz), Alesis iO26 transfers 4 data channels per data block in CIP. Both iO14 and iO26 have the same contents in their configuration ROM. For this reason, ALSA Dice driver attempts to distinguish them according to the value of TX0_AUDIO register at probe callback. Although the way is valid at lower and middle sampling rate, it's lastly invalid at higher sampling rate because because the two models returns the same value for read transaction to the register. In the most cases, users just plug-in the device and ALSA dice driver detects it. In the case, the device runs at lower sampling rate and the driver detects expectedly. For this reason, this commit leaves the way to detect as is. Fixes: 28b208f600a3 ("ALSA: dice: add parameters of stream formats for models produced by Alesis") Cc: <stable(a)vger.kernel.org> # v4.18+ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice-alesis.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/dice/dice-alesis.c b/sound/firewire/dice/dice-alesis.c index 218292bdace6..f5b325263b67 100644 --- a/sound/firewire/dice/dice-alesis.c +++ b/sound/firewire/dice/dice-alesis.c @@ -15,7 +15,7 @@ alesis_io14_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { static const unsigned int alesis_io26_tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT] = { - {10, 10, 8}, /* Tx0 = Analog + S/PDIF. */ + {10, 10, 4}, /* Tx0 = Analog + S/PDIF. */ {16, 8, 0}, /* Tx1 = ADAT1 + ADAT2. */ }; -- 2.20.1

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[alsa-devel] [PATCH 0/6] soundwire: inits and PM additions for 5.4
by Pierre-Louis Bossart 16 Sep '19

16 Sep '19

This is an update on the RFC, to be applied after the '[PATCH v2 0/3] soundwire: debugfs support for 5.4' and '[PATCH 00/17] soundwire: fixes for 5.4' series. Total that makes 28 patches submitted for review, broken in 3 sets. Changes since RFC (Feedback from GregKH, Vinod, Cezary, Guennadi): Squashed init sequence fixes in one patch, which remains readable. Tested all return values and called update_config() as needed. Fixed hw-reset debugfs (removed -unsafe and noisy dev_info traces) Simplified enable_interrupt() with goto Fixed style, removed typos and FIXMES in pm_runtime code Clarified commit messages Pierre-Louis Bossart (6): soundwire: fix startup sequence for Intel/Cadence soundwire: cadence_master: add hw_reset capability in debugfs soundwire: intel: add helper for initialization soundwire: intel: Add basic power management support soundwire: cadence_master: make clock stop exit configurable on init soundwire: intel: add pm_runtime support drivers/soundwire/cadence_master.c | 135 ++++++++++++++------ drivers/soundwire/cadence_master.h | 5 +- drivers/soundwire/intel.c | 194 +++++++++++++++++++++++++++-- 3 files changed, 289 insertions(+), 45 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH v4] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 16 Sep '19

16 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

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[alsa-devel] [GIT PULL] ASoC updates for v5.4 round II
by Mark Brown 16 Sep '19

16 Sep '19

The following changes since commit 6fa5963c37a2e3335eba0b7455e35a01318ebc15: ASoC: hdmi-codec: Add an op to set callback function for plug event (2019-09-09 14:50:03 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.4-2 for you to fetch changes up to a54f8485727e6e85b5dfe1d9708b80d13fe8bf5f: Merge branch 'asoc-5.4' into asoc-next (2019-09-15 10:31:44 +0100) ---------------------------------------------------------------- ASoC: Final merge window fixes for v5.4 A few small fixes and one feature that came in since I sent you the earlier pull request. ---------------------------------------------------------------- Mark Brown (2): Merge branch 'asoc-5.3' into asoc-linus Merge branch 'asoc-5.4' into asoc-next Peter Ujfalusi (1): ASoC: dmaengine: Replace strncpy() with strscpy_pad() for pcm->name Saiyam Doshi (2): ASoC: wcd9335: remove redundant use of ret variable ASoC: sdm845: remove unneeded semicolon Viorel Suman (1): ASoC: fsl_sai: Implement set_bclk_ratio .clang-format | 17 +- .gitignore | 3 + .mailmap | 8 + Documentation/PCI/index.rst | 2 +- Documentation/PCI/pci-error-recovery.rst | 5 +- .../PCI/{picebus-howto.rst => pciebus-howto.rst} | 0 Documentation/RCU/rculist_nulls.txt | 2 +- Documentation/admin-guide/conf.py | 10 - Documentation/admin-guide/hw-vuln/spectre.rst | 88 ++- Documentation/admin-guide/kernel-parameters.txt | 17 +- Documentation/admin-guide/mm/transhuge.rst | 2 +- Documentation/admin-guide/sysctl/net.rst | 29 +- Documentation/conf.py | 30 +- Documentation/core-api/conf.py | 10 - Documentation/crypto/conf.py | 10 - Documentation/dev-tools/conf.py | 10 - Documentation/devicetree/bindings/Makefile | 4 +- .../devicetree/bindings/arm/idle-states.txt | 2 +- Documentation/devicetree/bindings/arm/renesas.yaml | 2 +- .../bindings/arm/socionext/milbeaut.yaml | 2 +- .../devicetree/bindings/arm/ti/ti,davinci.yaml | 2 +- .../bindings/clock/allwinner,sun4i-a10-ccu.yaml | 2 +- .../intel,ixp4xx-network-processing-engine.yaml | 2 +- .../devicetree/bindings/iio/accel/adi,adxl345.yaml | 2 +- .../devicetree/bindings/iio/accel/adi,adxl372.yaml | 2 +- .../interrupt-controller/amazon,al-fic.txt | 16 +- .../intel,ixp4xx-interrupt.yaml | 2 +- .../interrupt-controller/snps,archs-idu-intc.txt | 30 +- ...er.yaml => intel,ixp4xx-ahb-queue-manager.yaml} | 2 +- .../bindings/net/allwinner,sun8i-a83t-emac.yaml | 2 +- Documentation/devicetree/bindings/net/dsa/ksz.txt | 1 + Documentation/devicetree/bindings/net/fsl-fec.txt | 30 +- Documentation/devicetree/bindings/net/macb.txt | 4 +- .../bindings/nvmem/allwinner,sun4i-a10-sid.yaml | 4 +- .../devicetree/bindings/nvmem/nvmem-consumer.yaml | 45 ++ Documentation/devicetree/bindings/nvmem/nvmem.txt | 81 +-- Documentation/devicetree/bindings/nvmem/nvmem.yaml | 93 +++ .../phy/allwinner,sun6i-a31-mipi-dphy.yaml | 2 +- .../bindings/pinctrl/st,stm32-pinctrl.yaml | 3 +- Documentation/devicetree/bindings/riscv/cpus.txt | 162 ----- 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| 4 +- arch/s390/include/asm/bitops.h | 73 +- arch/s390/include/asm/page.h | 2 + arch/s390/include/asm/qdio.h | 10 +- arch/s390/include/asm/setup.h | 5 +- arch/s390/include/asm/unistd.h | 1 + arch/s390/include/uapi/asm/bpf_perf_event.h | 2 +- arch/s390/include/uapi/asm/ipl.h | 2 +- arch/s390/include/uapi/asm/zcrypt.h | 35 +- arch/s390/kernel/dumpstack.c | 6 +- arch/s390/kernel/head64.S | 7 - arch/s390/kernel/ipl.c | 9 - arch/s390/kernel/machine_kexec_reloc.c | 1 + arch/s390/kernel/perf_cpum_cf_diag.c | 2 +- arch/s390/kernel/setup.c | 3 +- arch/s390/kernel/syscalls/syscall.tbl | 2 +- arch/s390/kernel/vdso.c | 5 - arch/s390/kernel/vmlinux.lds.S | 10 +- arch/s390/kvm/kvm-s390.c | 10 - arch/s390/lib/xor.c | 1 + arch/s390/mm/dump_pagetables.c | 12 +- arch/s390/mm/fault.c | 3 + arch/s390/mm/gmap.c | 4 +- arch/s390/mm/pgalloc.c | 6 +- arch/s390/net/bpf_jit_comp.c | 12 +- arch/s390/scripts/Makefile.chkbss | 3 +- arch/sh/include/uapi/asm/setup.h | 2 +- arch/sh/include/uapi/asm/types.h | 2 +- arch/sh/kernel/disassemble.c | 5 +- arch/sh/kernel/hw_breakpoint.c | 1 + arch/sparc/include/uapi/asm/oradax.h | 2 +- arch/um/include/shared/timer-internal.h | 14 +- arch/um/kernel/process.c | 2 +- arch/um/kernel/time.c | 16 +- arch/x86/Makefile | 1 + arch/x86/boot/compressed/pgtable_64.c | 13 +- arch/x86/boot/string.c | 8 + arch/x86/entry/calling.h | 17 + arch/x86/entry/entry_32.S | 13 +- arch/x86/entry/entry_64.S | 21 +- arch/x86/events/amd/ibs.c | 13 +- arch/x86/events/core.c | 2 +- arch/x86/events/intel/core.c | 15 +- arch/x86/events/intel/ds.c | 2 +- arch/x86/hyperv/mmu.c | 8 +- arch/x86/include/asm/bootparam_utils.h | 64 +- arch/x86/include/asm/cpufeatures.h | 3 + arch/x86/include/asm/ftrace.h | 1 - arch/x86/include/asm/intel-family.h | 15 + arch/x86/include/asm/kvm_host.h | 10 +- arch/x86/include/asm/msr-index.h | 1 + arch/x86/include/asm/nospec-branch.h | 2 +- arch/x86/include/asm/perf_event.h | 12 +- arch/x86/include/asm/uaccess.h | 4 +- arch/x86/include/asm/vdso/gettimeofday.h 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| 4 +- .../amd/display/dc/clk_mgr/dce112/dce112_clk_mgr.c | 4 +- .../amd/display/dc/clk_mgr/dce120/dce120_clk_mgr.c | 4 +- .../amd/display/dc/clk_mgr/dcn20/dcn20_clk_mgr.c | 3 + drivers/gpu/drm/amd/display/dc/core/dc.c | 17 +- drivers/gpu/drm/amd/display/dc/core/dc_link.c | 30 +- drivers/gpu/drm/amd/display/dc/core/dc_link_dp.c | 39 +- drivers/gpu/drm/amd/display/dc/core/dc_resource.c | 11 +- drivers/gpu/drm/amd/display/dc/core/dc_stream.c | 3 +- drivers/gpu/drm/amd/display/dc/dce/dce_abm.c | 4 + .../amd/display/dc/dce110/dce110_hw_sequencer.c | 24 +- .../drm/amd/display/dc/dcn10/dcn10_hw_sequencer.c | 21 +- .../gpu/drm/amd/display/dc/dcn10/dcn10_resource.c | 2 +- drivers/gpu/drm/amd/display/dc/dcn20/dcn20_dccg.c | 5 + .../gpu/drm/amd/display/dc/dcn20/dcn20_hubbub.c | 18 +- drivers/gpu/drm/amd/display/dc/dcn20/dcn20_hwseq.c | 22 +- drivers/gpu/drm/amd/display/dc/dcn20/dcn20_optc.c | 2 +- .../gpu/drm/amd/display/dc/dcn20/dcn20_resource.c | 4 + 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| 12 + drivers/gpu/drm/i915/gt/intel_gt_pm.c | 21 +- drivers/gpu/drm/i915/gt/intel_gt_pm.h | 2 +- drivers/gpu/drm/i915/gt/intel_lrc.c | 10 +- drivers/gpu/drm/i915/gt/intel_reset.c | 58 +- drivers/gpu/drm/i915/gt/intel_ringbuffer.c | 31 +- drivers/gpu/drm/i915/gt/intel_workarounds.c | 38 +- drivers/gpu/drm/i915/gt/mock_engine.c | 1 + drivers/gpu/drm/i915/gt/selftest_reset.c | 5 +- drivers/gpu/drm/i915/gt/selftest_workarounds.c | 7 +- drivers/gpu/drm/i915/gvt/cmd_parser.c | 10 - drivers/gpu/drm/i915/gvt/fb_decoder.c | 6 +- drivers/gpu/drm/i915/gvt/gtt.c | 9 + drivers/gpu/drm/i915/gvt/kvmgt.c | 12 + drivers/gpu/drm/i915/gvt/scheduler.c | 59 +- drivers/gpu/drm/i915/gvt/trace_points.c | 2 - drivers/gpu/drm/i915/i915_drv.c | 6 + drivers/gpu/drm/i915/i915_drv.h | 5 +- drivers/gpu/drm/i915/i915_gem.c | 25 +- drivers/gpu/drm/i915/i915_gem_gtt.c | 8 +- drivers/gpu/drm/i915/i915_gpu_error.c | 7 +- drivers/gpu/drm/i915/i915_perf.c | 67 +- drivers/gpu/drm/i915/i915_trace.h | 76 +-- 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kernel/trace/trace_functions_graph.c | 17 +- kernel/trace/trace_probe.c | 3 +- lib/Kconfig.kasan | 11 +- lib/Makefile | 3 +- lib/dim/dim.c | 4 +- lib/dim/net_dim.c | 56 ++ lib/kfifo.c | 3 +- lib/logic_pio.c | 73 +- lib/raid6/Makefile | 2 +- lib/test_firmware.c | 5 +- lib/test_meminit.c | 2 +- lib/vdso/gettimeofday.c | 79 ++- mm/Makefile | 1 + mm/balloon_compaction.c | 72 +- mm/compaction.c | 11 +- mm/hmm.c | 10 +- mm/huge_memory.c | 55 +- mm/hugetlb.c | 19 + mm/kasan/common.c | 10 +- mm/kmemleak.c | 4 +- mm/memcontrol.c | 146 +++- mm/memory_hotplug.c | 2 - mm/mempolicy.c | 134 ++-- {kernel => mm}/memremap.c | 30 + mm/migrate.c | 21 +- mm/page_alloc.c | 19 +- mm/rmap.c | 8 + mm/shmem.c | 2 +- mm/slub.c | 8 +- mm/usercopy.c | 2 +- mm/vmalloc.c | 21 +- mm/vmscan.c | 27 +- mm/workingset.c | 10 +- mm/z3fold.c | 104 ++- mm/zsmalloc.c | 80 ++- net/batman-adv/bat_iv_ogm.c | 20 +- net/batman-adv/bat_v_ogm.c | 18 +- net/batman-adv/multicast.c | 8 +- net/batman-adv/netlink.c | 2 +- net/bluetooth/hci_core.c | 1 + net/bluetooth/hci_debugfs.c | 31 + net/bluetooth/hidp/core.c | 9 +- net/bluetooth/l2cap_core.c | 2 +- net/bridge/br.c | 5 +- net/bridge/br_multicast.c | 3 + net/bridge/br_private.h | 9 +- net/bridge/br_vlan.c | 29 +- net/bridge/netfilter/Kconfig | 6 +- net/bridge/netfilter/ebtables.c | 40 +- net/bridge/netfilter/nft_meta_bridge.c | 12 +- net/can/gw.c | 48 +- net/ceph/crypto.c | 6 +- net/ceph/osd_client.c | 9 +- net/core/dev.c | 17 +- net/core/filter.c | 14 +- net/core/flow_dissector.c | 2 +- net/core/flow_offload.c | 22 +- net/core/netpoll.c | 6 +- net/core/skmsg.c | 4 +- net/core/sock.c | 50 +- net/core/sock_diag.c | 3 +- net/core/sock_map.c | 19 +- net/core/stream.c | 16 +- net/dsa/slave.c | 6 +- net/dsa/switch.c | 3 + net/dsa/tag_8021q.c | 2 + net/dsa/tag_sja1105.c | 12 +- net/ieee802154/6lowpan/reassembly.c | 2 +- net/ieee802154/socket.c | 2 +- net/ipv4/fib_trie.c | 2 +- net/ipv4/icmp.c | 10 +- net/ipv4/igmp.c | 4 +- net/ipv4/inet_fragment.c | 41 +- 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delete mode 100644 arch/arm/mach-netx/nxdkn.c delete mode 100644 arch/arm/mach-netx/nxeb500hmi.c delete mode 100644 arch/arm/mach-netx/pfifo.c delete mode 100644 arch/arm/mach-netx/time.c delete mode 100644 arch/arm/mach-netx/xc.c rename arch/parisc/configs/{default_defconfig => defconfig} (100%) delete mode 100644 arch/riscv/lib/udivdi3.S create mode 100644 arch/s390/boot/version.c delete mode 100644 arch/x86/purgatory/string.c rename {include/misc => drivers/auxdisplay}/charlcd.h (94%) delete mode 100644 drivers/tty/serial/netx-serial.c rename {kernel => mm}/memremap.c (90%) create mode 100755 tools/testing/selftests/bpf/test_xdp_vlan_mode_generic.sh create mode 100755 tools/testing/selftests/bpf/test_xdp_vlan_mode_native.sh create mode 100644 tools/testing/selftests/kvm/config

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[alsa-devel] fix memory leak in snd_pcm_set_chmap
by GitHub pull_request - opened 16 Sep '19

16 Sep '19

alsa-project/alsa-lib pull request #11 was opened from conradjones: memory leak in snd_pcm_set_chmap : memory is allocated in the function pointed at by : pcm->ops->get_chmap called in: snd_pcm_chmap_t *snd_pcm_get_chmap(snd_pcm_t *pcm) Request URL : https://github.com/alsa-project/alsa-lib/pull/11 Patch URL : https://github.com/alsa-project/alsa-lib/pull/11.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] fix memory leak in snd_pcm_set_chmap
by GitHub pull_request - edited 16 Sep '19

16 Sep '19

alsa-project/alsa-lib pull request #10 was edited from conradjones: ignore Request URL : https://github.com/alsa-project/alsa-lib/pull/10 Patch URL : https://github.com/alsa-project/alsa-lib/pull/10.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] fix memory leak in snd_pcm_set_chmap
by GitHub pull_request - opened 15 Sep '19

15 Sep '19

alsa-project/alsa-lib pull request #10 was opened from conradjones: asan detects this as a memory leak. snd_pcm_get_chmap clearly states that the caller is required to free the returned value, i cannot see it stored anywhere so cannot see how it could be freed. Request URL : https://github.com/alsa-project/alsa-lib/pull/10 Patch URL : https://github.com/alsa-project/alsa-lib/pull/10.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Add Hiby device family to quirks for native DSD support
by ilya.pshonkin＠netforce.ua 15 Sep '19

15 Sep '19

From: Sudokamikaze <sudokamikaze(a)protonmail.com> This patch adds quirk VID ID for Hiby portable players family with native DSD playback support Signed-off-by: Sudokamikaze <sudokamikaze(a)protonmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 78858918cbc1..64a8d73972e3 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1655,6 +1655,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x152a: /* Thesycon devices */ case 0x25ce: /* Mytek devices */ case 0x2ab6: /* T+A devices */ + case 0xc502: /* HiBy devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.23.0

1 0

[alsa-devel] Jack detection in Lenovo OneLink+ and other docks
by Benjamin Berg 15 Sep '19

15 Sep '19

Hi, I have a Lenovo OneLink+ and TB3 docks here. The USB audio adapter on these docks work fine. However, Jack detection is not functional unfortunately which is somewhat annoying. There are HID devices available on the docking stations which seem to report the jack sensing information. Assuming we can decode this information correctly, does anyone have an idea how this can be hooked up so that the information will be available on the audio device itself? Benjamin The OneLink+ dock exposes the following interfaces: |__ Port 1: Dev 90, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 3: Dev 92, If 1, Class=CDC Data, Driver=cdc_ether, 480M |__ Port 3: Dev 92, If 0, Class=Communications, Driver=cdc_ether, 480M |__ Port 4: Dev 94, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 4: Dev 94, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 4: Dev 94, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 94, If 1, Class=Audio, Driver=snd-usb-audio, 12M Similar on the TB 3 Dock from Lenovo: /: Bus 05.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 1: Dev 2, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 2, If 1, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 2, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 2, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 3, If 0, Class=Human Interface Device, Driver=usbhid, 12M On each of these devices, plugging/unplugging will fire various HID events, seemingly with some sensing information. On the OneLink Plus: D: 0 R: 69 05 0c 09 01 a1 01 85 01 15 00 25 01 09 e9 09 ea 75 01 95 02 81 42 09 e2 95 01 81 06 06 01 ff 09 01 95 09 81 06 05 0c 09 cd 09 b7 09 b5 09 b6 95 04 81 06 06 07 ff 09 01 75 08 95 0d 81 06 09 00 95 0f 91 02 c0 N: C-Media Electronics Inc. ThinkPad OneLink Plus Dock Audio P: usb-0000:00:14.0-4.1.4/input3 I: 3 17ef 3055 # Plug Headphone D: 0 E: 0.000000 16 01 00 06 00 00 20 83 ab 08 00 08 c0 10 00 00 00 E: 3.103810 16 01 00 0a 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Unplug headphone E: 9.951613 16 01 00 08 00 00 20 83 ab 08 00 03 80 00 00 00 00 E: 10.016073 16 01 00 04 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Plug Microphone E: 17.824186 16 01 00 06 00 00 20 83 ab 08 00 03 80 00 00 00 00 E: 20.928234 16 01 00 0a 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Unplug Microphone E: 26.734048 16 01 00 08 00 00 20 83 ab 08 00 08 c0 10 00 00 00 E: 26.784007 16 01 00 04 00 00 20 83 ab 08 00 03 80 10 00 00 00 # Plug headset E: 44.160216 16 01 00 06 00 00 20 83 ab 08 00 03 80 00 00 00 00 # Unplug headset E: 56.687367 16 01 00 04 00 00 20 83 ab 08 00 08 c0 10 00 00 00 On the TB 3 dock: D: 0 R: 120 05 0c 09 01 a1 01 85 01 15 00 25 01 09 e9 09 ea 75 01 95 02 81 02 09 e2 95 01 81 06 09 00 95 02 81 03 09 36 a1 02 05 09 19 01 29 02 75 02 95 01 15 01 25 02 81 40 c0 05 0c 09 00 15 00 25 01 75 01 95 01 81 03 85 02 05 0c 09 00 95 10 81 02 85 03 09 00 91 02 85 04 09 00 75 08 95 26 91 02 85 05 09 00 95 22 81 02 85 06 09 00 95 24 91 02 85 07 09 00 95 20 81 02 c0 N: Lenovo ThinkPad Thunderbolt 3 Dock USB Audio P: usb-0000:0a:00.0-1/input3 I: 3 17ef 306a # Plug in headphone D: 0 E: 0.000000 3 02 00 80 # Unplug headphone E: 16.480011 3 02 00 00 # Plug in microphone E: 34.224331 3 02 00 80 # Unplug microphone E: 46.960128 3 02 00 00 # Plug in headset E: 62.592182 3 02 00 80 E: 63.120449 3 02 00 c0 # Unplug headset E: 72.608250 3 02 00 80 E: 72.624253 3 02 00 00

2 1

[alsa-devel] Applied "ASoC: sdm845: remove unneeded semicolon" to the asoc tree
by Mark Brown 15 Sep '19

15 Sep '19

The patch ASoC: sdm845: remove unneeded semicolon has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.4 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From fca11622d600228bec405456f41590155b3a3eca Mon Sep 17 00:00:00 2001 From: Saiyam Doshi <saiyamdoshi.in(a)gmail.com> Date: Sat, 14 Sep 2019 08:41:33 +0530 Subject: [PATCH] ASoC: sdm845: remove unneeded semicolon Remove excess semicolon after closing parenthesis. Signed-off-by: Saiyam Doshi <saiyamdoshi.in(a)gmail.com> Link: https://lore.kernel.org/r/20190914031133.GA28447@SD Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 882f52ed8231..28f3cef696e6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -319,7 +319,7 @@ static void sdm845_snd_shutdown(struct snd_pcm_substream *substream) snd_soc_dai_set_sysclk(cpu_dai, Q6AFE_LPASS_CLK_ID_PRI_MI2S_IBIT, 0, SNDRV_PCM_STREAM_PLAYBACK); - }; + } break; case SECONDARY_MI2S_TX: -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Add dmic format constraint
by Yu-Hsuan Hsu 14 Sep '19

14 Sep '19

24 bits recording from DMIC is not supported for KBL platform because the TDM slot between PCH and codec is 16 bits only. We should add a constraint to remove that unsupported format. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda8784f1a01..67b276a65a8d2d 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -400,6 +400,9 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, dmic_constraints); + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -- 2.23.0.162.g0b9fbb3734-goog

2 5

[alsa-devel] [PATCH] ASoC: fsl_sai: Implement set_bclk_ratio
by Daniel Baluta 13 Sep '19

13 Sep '19

From: Viorel Suman <viorel.suman(a)nxp.com> This is to allow machine drivers to set a certain bitclk rate which might not be exactly rate * frame size. Cc: NXP Linux Team <linux-imx(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 21 +++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fe126029f4e3..e896b577b1f7 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -137,6 +137,16 @@ static int fsl_sai_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, return 0; } +static int fsl_sai_set_dai_bclk_ratio(struct snd_soc_dai *dai, + unsigned int ratio) +{ + struct fsl_sai *sai = snd_soc_dai_get_drvdata(dai); + + sai->bclk_ratio = ratio; + + return 0; +} + static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -423,8 +433,14 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, slot_width = sai->slot_width; if (!sai->is_slave_mode) { - ret = fsl_sai_set_bclk(cpu_dai, tx, - slots * slot_width * params_rate(params)); + if (sai->bclk_ratio) + ret = fsl_sai_set_bclk(cpu_dai, tx, + sai->bclk_ratio * + params_rate(params)); + else + ret = fsl_sai_set_bclk(cpu_dai, tx, + slots * slot_width * + params_rate(params)); if (ret) return ret; @@ -640,6 +656,7 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { + .set_bclk_ratio = fsl_sai_set_dai_bclk_ratio, .set_sysclk = fsl_sai_set_dai_sysclk, .set_fmt = fsl_sai_set_dai_fmt, .set_tdm_slot = fsl_sai_set_dai_tdm_slot, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 3a3f6f8e5595..f96f8d97489d 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -177,6 +177,7 @@ struct fsl_sai { unsigned int mclk_streams; unsigned int slots; unsigned int slot_width; + unsigned int bclk_ratio; const struct fsl_sai_soc_data *soc_data; struct snd_dmaengine_dai_dma_data dma_params_rx; -- 2.17.1

4 6

[alsa-devel] [PATCH v3] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 13 Sep '19

13 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Aduio Sample Channel Status information in the IEC60958 frame. The status bit is already available in iec.status of hdmi_codec_params. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Note that these registers are only for interfaces: I2S audio interface, General Purpose Audio (GPA), or AHB audio DMA (AHBAUDDMA). For S/PDIF interface this information comes from the stream. Currently this function dw_hdmi_set_channel_status is only called from dw-hdmi-i2s-audio in I2S setup. Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Change from v2 to v3: 1. Reuse what is already set in iec.status in hw_param. 2. Remove all useless definition of registers and values. 3. Note that the original sampling frequency is not written to the channel status as we reuse create_iec958_consumer in pcm_iec958.c. Without that it can still play audio fine. .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 1 + drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 20 +++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 2 ++ include/drm/bridge/dw_hdmi.h | 1 + 4 files changed, 24 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c index 34d8e837555f..20f4f92dd866 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c @@ -102,6 +102,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, } dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); + dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); dw_hdmi_set_channel_allocation(hdmi, hparms->cea.channel_allocation); diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..aa7efd4da1c8 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,26 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +/* + * When transmitting IEC60958 linear PCM audio, these registers allow to + * configure the channel status information of all the channel status + * bits in the IEC60958 frame. For the moment this configuration is only + * used when the I2S audio interface, General Purpose Audio (GPA), + * or AHB audio DMA (AHBAUDDMA) interface is active + * (for S/PDIF interface this information comes from the stream). + */ +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, + u8 *channel_status) +{ + /* + * Set channel status register for frequency and word length. + * Use default values for other registers. + */ + hdmi_writeb(hdmi, channel_status[3], HDMI_FC_AUDSCHNLS7); + hdmi_writeb(hdmi, channel_status[4], HDMI_FC_AUDSCHNLS8); +} +EXPORT_SYMBOL_GPL(dw_hdmi_set_channel_status); + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..fcff5059db24 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,8 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 diff --git a/include/drm/bridge/dw_hdmi.h b/include/drm/bridge/dw_hdmi.h index cf528c289857..4b3e863c4f8a 100644 --- a/include/drm/bridge/dw_hdmi.h +++ b/include/drm/bridge/dw_hdmi.h @@ -156,6 +156,7 @@ void dw_hdmi_setup_rx_sense(struct dw_hdmi *hdmi, bool hpd, bool rx_sense); void dw_hdmi_set_sample_rate(struct dw_hdmi *hdmi, unsigned int rate); void dw_hdmi_set_channel_count(struct dw_hdmi *hdmi, unsigned int cnt); +void dw_hdmi_set_channel_status(struct dw_hdmi *hdmi, u8 *channel_status); void dw_hdmi_set_channel_allocation(struct dw_hdmi *hdmi, unsigned int ca); void dw_hdmi_audio_enable(struct dw_hdmi *hdmi); void dw_hdmi_audio_disable(struct dw_hdmi *hdmi); -- 2.23.0.162.g0b9fbb3734-goog

4 6

[alsa-devel] [PATCH 0/3] update supported sample format
by Shengjiu Wang 13 Sep '19

13 Sep '19

update supported sample format Shengjiu Wang (3): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++++--------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 93 +++++++++++++++++++++++++++++++++--- 3 files changed, 135 insertions(+), 30 deletions(-) -- 2.21.0

4 12

[alsa-devel] [PATCH 0/2] ASoC: mt8183: fix audio playback slowly after playback during bootup
by Jiaxin Yu 12 Sep '19

12 Sep '19

These two patches fix audio playack slowly after playbakc during bootup. The patches are based on broonie tree "for-next" branch. Jiaxin Yu (2): ASoC: dt-bingdings: mediatek: mt8183: add a property "medaitek,toprgu" ASoC: mt8183: fix audio playback slowly after playback during bootup .../bindings/sound/mt8183-afe-pcm.txt | 2 ++ sound/soc/mediatek/common/mtk-base-afe.h | 1 + sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 21 +++++++++++++++++++ sound/soc/mediatek/mt8183/mt8183-reg.h | 6 ++++++ 4 files changed, 30 insertions(+) -- 2.18.0

2 4

[alsa-devel] [PATCH v3 00/10] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 12 Sep '19

12 Sep '19

Hi all, here's the third 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. This is now first series without the RFC tag. The series has all machine drivers converted that are directly impacted. v3 changes: - Merged two mst_no_extra_pcms patches (Takashi's feedback). - Use mach_params instead of module name lookups to determine which codec driver is used at runtime (Pierre's feedback). - Extended machine driver support to four more drivers. This includes now all machine drivers which are used with SOF and which support HDMI audio. With this patchset, the machine drivers continue to work with both SOF and SST DSP drivers, and can be configured to use hdac-hdmi also with SOF (via kernel module parameter -> this is for distros to provide backwards compatibility for UCM files). - Various smaller cleanups across the patchset. 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 (10): ALSA: hda/hdmi - implement mst_no_extra_pcms flag ALSA: hda/hdmi - allow control creation without a linked pcm ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ASoC: SOF: Intel: add support for snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: Intel: skl-hda-dsp-generic: fix include guard name 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 | 23 +++-- sound/soc/codecs/hdac_hda.c | 95 ++++++++++++++++--- sound/soc/codecs/hdac_hda.h | 12 ++- 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 | 87 +++++++++++++++++ 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 | 19 +++- sound/soc/sof/intel/hda.c | 6 ++ sound/soc/sof/intel/hda.h | 6 +- 17 files changed, 315 insertions(+), 28 deletions(-) create mode 100644 sound/soc/intel/boards/hda_dsp_common.h -- 2.17.1

2 20

[alsa-devel] [PATCH v3] ASoC: Intel: Skylake: Recover BXT FW on DSP boot timeout error
by Pawel Harlozinski 12 Sep '19

12 Sep '19

When DSP boots with timeout error try to reload firmware to keep best user expeirience. Change-Id: Ic7377da687e7441a487c443f9de1ebae5967afa2 Signed-off-by: Szymon Mielczarek <szymonx.mielczarek(a)intel.com> Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- sound/soc/intel/skylake/bxt-sst.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/bxt-sst.c b/sound/soc/intel/skylake/bxt-sst.c index 92a82e6b5fe6..fdf4c198445e 100644 --- a/sound/soc/intel/skylake/bxt-sst.c +++ b/sound/soc/intel/skylake/bxt-sst.c @@ -446,14 +446,19 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) /* If core 1 was turned on for booting core 0, turn it off */ skl_dsp_core_power_down(ctx, SKL_DSP_CORE_MASK(1)); + /* FIXME this FW recovery applies only for BXT */ if (ret == 0) { - dev_err(ctx->dev, "%s: DSP boot timeout\n", __func__); - dev_err(ctx->dev, "Error code=0x%x: FW status=0x%x\n", + dev_warn(ctx->dev, + "DSP boot timeout: Error code=0x%x: FW status=0x%x. Reloading FW\n", sst_dsp_shim_read(ctx, BXT_ADSP_ERROR_CODE), sst_dsp_shim_read(ctx, BXT_ADSP_FW_STATUS)); - dev_err(ctx->dev, "Failed to set core0 to D0 state\n"); - ret = -EIO; - goto err; + + ret = bxt_sst_init_fw(skl->dev, skl); + if (ret < 0) { + dev_err(ctx->dev, "Reload fw failed: %d. Failed to set core0 to D0 state\n", + ret); + goto err; + } } } @@ -478,7 +483,6 @@ static int bxt_set_dsp_D0(struct sst_dsp *ctx, unsigned int core_id) if (core_id == SKL_DSP_CORE0_ID) core_mask |= SKL_DSP_CORE_MASK(1); skl_dsp_disable_core(ctx, core_mask); - return ret; } -- 2.17.1

2 1

[alsa-devel] [PATCH v3] tas2770: add tas2770 smart PA kernel driver
by shifu0704＠thundersoft.com 12 Sep '19

12 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> add tas2770 smart PA kernel driver Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2770.c | 952 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2770.h | 166 ++++++++ 4 files changed, 1125 insertions(+) create mode 100644 sound/soc/codecs/tas2770.c create mode 100644 sound/soc/codecs/tas2770.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8f3e787..cc92da3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -111,6 +111,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_STI_SAS select SND_SOC_TAS2552 if I2C + select SND_SOC_TAS2770 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C select SND_SOC_TFA9879 if I2C @@ -652,6 +653,10 @@ config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C +config SND_SOC_TAS2770 + tristate "Texas Instruments TAS2770 speaker amplifier" + depends on I2C + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5305cc6..63b8488 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -116,6 +116,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-sti-sas-objs := sti-sas.o snd-soc-tas5086-objs := tas5086.o snd-soc-tas571x-objs := tas571x.o +snd-soc-tas2770-objs := tas2770.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o @@ -332,6 +333,7 @@ obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o +obj-$(CONFIG_SND_SOC_TAS2770) += snd-soc-tas2770.o obj-$(CONFIG_SND_SOC_TFA9879) += snd-soc-tfa9879.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c new file mode 100644 index 0000000..65aeace --- /dev/null +++ b/sound/soc/codecs/tas2770.c @@ -0,0 +1,952 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2770 20-W Digital Input Mono Class-D +// Audio Amplifier with Speaker I/V Sense +// +// Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ +// Author: Tracy Yi <tracy-yi(a)ti.com> +// Frank Shi <shifu0704(a)thundersoft.com> + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/err.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/firmware.h> +#include <linux/regmap.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "tas2770.h" + +#define TAS2770_MDELAY 0xFFFFFFFE +#define TAS2770_CHECK_PERIOD 5000 /* 5 second */ + +static void tas2770_reset(struct tas2770_priv *tas2770) +{ + if (tas2770->reset_gpio) { + gpiod_set_value_cansleep(tas2770->reset_gpio, 0); + msleep(20); + gpiod_set_value_cansleep(tas2770->reset_gpio, 1); + } + snd_soc_component_write(tas2770->component, TAS2770_SW_RST, + TAS2770_RST); +} + +static int tas2770_regmap_read(struct tas2770_priv *tas2770, + unsigned int reg, unsigned int *value) +{ + int result = 0; + int retry_count = TAS2770_I2C_RETRY_COUNT; + + while (retry_count--) { + result = snd_soc_component_read(tas2770->component, reg, + value); + if (!result) + break; + + msleep(20); + } + if (!retry_count) + return -ETIMEDOUT; + + return 0; +} + +static int tas2770_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_ACTIVE); + tas2770->power_state = SND_SOC_BIAS_STANDBY; + break; + + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_MUTE); + tas2770->power_state = SND_SOC_BIAS_PREPARE; + break; + + case SND_SOC_BIAS_OFF: + snd_soc_component_update_bits(component, TAS2770_PWR_CTRL, + TAS2770_PWR_CTRL_MASK, + TAS2770_PWR_CTRL_SHUTDOWN); + tas2770->power_state = SND_SOC_BIAS_OFF; + break; + + default: + dev_err(tas2770->dev, + "wrong power level setting %d\n", level); + return -EINVAL; + } + + return 0; +} + +#ifdef CONFIG_PM +static int tas2770_codec_suspend(struct snd_soc_component *component) +{ + int ret; + + ret = tas2770_set_bias_level(component, SND_SOC_BIAS_OFF); + if (ret) + return -EINVAL; + + return 0; +} + +static int tas2770_codec_resume(struct snd_soc_component *component) +{ + int ret; + + ret = tas2770_set_bias_level(component, SND_SOC_BIAS_STANDBY); + if (ret) + return -EINVAL; + + return 0; +} +#else +#define tas2770_codec_suspend NULL +#define tas2770_codec_resume NULL +#endif + +static const char * const tas2770_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL( + tas2770_ASI1_src_enum, TAS2770_TDM_CFG_REG2, + 4, tas2770_ASI1_src); + +static const struct snd_kcontrol_new tas2770_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2770_ASI1_src_enum); + +static int tas2770_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + ret = tas2770_set_bias_level(component, + SND_SOC_BIAS_PREPARE); + if (ret) + goto end; + break; + case SND_SOC_DAPM_PRE_PMD: + ret = tas2770_set_bias_level(component, + SND_SOC_BIAS_OFF); + if (ret) + goto end; + break; + default: + dev_err(tas2770->dev, "Not supported evevt\n"); + return -EINVAL; + } + +end: + return ret; +} + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 3, 1, 1); +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2770_PWR_CTRL, 2, 1, 1); + +static const struct snd_soc_dapm_widget tas2770_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, + &tas2770_asi1_mux), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2770_PWR_CTRL, 3, 1, + &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2770_PWR_CTRL, 2, 1, + &vsense_switch), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2770_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON") +}; + +static const struct snd_soc_dapm_route tas2770_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + {"DAC", NULL, "ASI1 Sel"}, + {"OUT", NULL, "DAC"}, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static int tas2770_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + int ret; + + if (mute) + ret = tas2770_set_bias_level(component, SND_SOC_BIAS_PREPARE); + else + ret = tas2770_set_bias_level(component, SND_SOC_BIAS_STANDBY); + + return ret; +} + +static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) +{ + int ret; + struct snd_soc_component *component = tas2770->component; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_16BITS); + tas2770->v_sense_slot = tas2770->i_sense_slot + 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_24BITS); + tas2770->v_sense_slot = tas2770->i_sense_slot + 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXW_MASK, + TAS2770_TDM_CFG_REG2_RXW_32BITS); + tas2770->v_sense_slot = tas2770->i_sense_slot + 4; + break; + + default: + return -EINVAL; + } + + tas2770->channel_size = bitwidth; + + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG5, + TAS2770_TDM_CFG_REG5_VSNS_MASK | + TAS2770_TDM_CFG_REG5_50_MASK, + TAS2770_TDM_CFG_REG5_VSNS_ENABLE | + tas2770->v_sense_slot); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG6, + TAS2770_TDM_CFG_REG6_ISNS_MASK | + TAS2770_TDM_CFG_REG6_50_MASK, + TAS2770_TDM_CFG_REG6_ISNS_ENABLE | + tas2770->i_sense_slot); + +end: + return ret; +} + +static int tas2770_set_samplerate(struct tas2770_priv *tas2770, + int samplerate) +{ + int ret; + struct snd_soc_component *component = tas2770->component; + + switch (samplerate) { + case 48000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 44100: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); + if (ret) + goto end; + break; + case 96000: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 88200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); + break; + case 19200: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_48KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + if (ret) + goto end; + break; + case 17640: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_SMP_MASK, + TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); + if (ret) + goto end; + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG0, + TAS2770_TDM_CFG_REG0_31_MASK, + TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); + break; + default: + ret = -EINVAL; + } + +end: + if (!ret) + tas2770->sampling_rate = samplerate; + return ret; +} + +static int tas2770_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + int ret; + + ret = tas2770_set_bitwidth(tas2770, params_format(params)); + if (ret < 0) + goto end; + + + ret = tas2770_set_samplerate(tas2770, params_rate(params)); + +end: + return ret; +} + +static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int ret; + int value = 0; + struct snd_soc_component *component = dai->component; + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(tas2770->dev, "ASI format master is not found\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_RSING; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 |= TAS2770_TDM_CFG_REG1_RX_FALING; + break; + default: + dev_err(tas2770->dev, "ASI format Inverse is not found\n"); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_RX_MASK, + asi_cfg_1); + if (ret) + return ret; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case (SND_SOC_DAIFMT_I2S): + tdm_rx_start_slot = 1; + break; + case (SND_SOC_DAIFMT_DSP_A): + tdm_rx_start_slot = 0; + break; + case (SND_SOC_DAIFMT_DSP_B): + tdm_rx_start_slot = 1; + break; + case (SND_SOC_DAIFMT_LEFT_J): + tdm_rx_start_slot = 0; + break; + default: + dev_err(tas2770->dev, + "DAI Format is not found, fmt=0x%x\n", fmt); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, + TAS2770_TDM_CFG_REG1_MASK, + (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); + if (ret) + return ret; + + value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); + + tas2770->asi_format = fmt; + + return 0; +} + +static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, + unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + int left_slot, right_slot; + int ret; + + if (tx_mask == 0 || rx_mask != 0) + return -EINVAL; + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + left_slot = 0; + right_slot = 0; + } else { + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + } + + if (tx_mask != 0 || left_slot >= slots || right_slot >= slots) + return -EINVAL; + + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_30_MASK, + (left_slot << TAS2770_TDM_CFG_REG3_30_SHIFT)); + if (ret) + return ret; + ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, + TAS2770_TDM_CFG_REG3_RXS_MASK, + (right_slot << TAS2770_TDM_CFG_REG3_RXS_SHIFT)); + if (ret) + return ret; + + switch (slot_width) { + case 16: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_16BITS); + break; + + case 24: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_24BITS); + break; + + case 32: + ret = snd_soc_component_update_bits(component, + TAS2770_TDM_CFG_REG2, + TAS2770_TDM_CFG_REG2_RXS_MASK, + TAS2770_TDM_CFG_REG2_RXS_32BITS); + break; + + case 0: + /* Do not change slot width */ + ret = 0; + break; + + default: + ret = -EINVAL; + } + + if (!ret) + tas2770->slot_width = slot_width; + + return ret; +} + +static struct snd_soc_dai_ops tas2770_dai_ops = { + .digital_mute = tas2770_mute, + .hw_params = tas2770_hw_params, + .set_fmt = tas2770_set_fmt, + .set_tdm_slot = tas2770_set_dai_tdm_slot, +}; + +#define TAS2770_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +#define TAS2770_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_192000\ + ) + +static struct snd_soc_dai_driver tas2770_dai_driver[] = { + { + .name = "tas2770 ASI1", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = TAS2770_RATES, + .formats = TAS2770_FORMATS, + }, + .ops = &tas2770_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int tas2770_codec_probe(struct snd_soc_component *component) +{ + struct tas2770_priv *tas2770 = + snd_soc_component_get_drvdata(component); + + tas2770->component = component; + + return 0; +} + +static DECLARE_TLV_DB_SCALE(tas2770_digital_tlv, 1100, 50, 0); +static DECLARE_TLV_DB_SCALE(tas2770_playback_volume, -12750, 50, 0); + +static const struct snd_kcontrol_new tas2770_snd_controls[] = { + SOC_SINGLE_TLV("Amp Output Gain", TAS2770_PLAY_CFG_REG0, + 0, 0x14, 0, + tas2770_digital_tlv), + SOC_SINGLE_TLV("Playback Volume", TAS2770_PLAY_CFG_REG2, + 0, TAS2770_PLAY_CFG_REG2_VMAX, 1, + tas2770_playback_volume), +}; + +static const struct snd_soc_component_driver soc_component_driver_tas2770 = { + .probe = tas2770_codec_probe, + .suspend = tas2770_codec_suspend, + .resume = tas2770_codec_resume, + .set_bias_level = tas2770_set_bias_level, + .controls = tas2770_snd_controls, + .num_controls = ARRAY_SIZE(tas2770_snd_controls), + .dapm_widgets = tas2770_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2770_dapm_widgets), + .dapm_routes = tas2770_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2770_audio_map), + .idle_bias_on = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static int tas2770_register_codec(struct tas2770_priv *tas2770) +{ + return devm_snd_soc_register_component(tas2770->dev, + &soc_component_driver_tas2770, + tas2770_dai_driver, ARRAY_SIZE(tas2770_dai_driver)); +} + +static void tas2770_load_config(struct tas2770_priv *tas2770) +{ + int ret; + struct snd_soc_component *component = tas2770->component; + + tas2770_reset(tas2770); + snd_soc_component_write(component, TAS2770_SW_RST, + TAS2770_RST); + + ret = tas2770_set_bitwidth(tas2770, tas2770->channel_size); + if (ret) + goto error_out; + + ret = tas2770_set_samplerate(tas2770, tas2770->sampling_rate); + if (ret) + goto error_out; + + ret = tas2770_set_bias_level(component, tas2770->power_state); + if (ret) + goto error_out; + +error_out: + return; +} + +static const struct reg_default tas2770_reg_defaults[] = { + { TAS2770_PAGE, 0x00 }, + { TAS2770_SW_RST, 0x00 }, + { TAS2770_PWR_CTRL, 0x0e }, + { TAS2770_PLAY_CFG_REG0, 0x10 }, + { TAS2770_PLAY_CFG_REG1, 0x01 }, + { TAS2770_PLAY_CFG_REG2, 0x00 }, + { TAS2770_MSC_CFG_REG0, 0x07 }, + { TAS2770_TDM_CFG_REG1, 0x02 }, + { TAS2770_TDM_CFG_REG2, 0x0a }, + { TAS2770_TDM_CFG_REG3, 0x10 }, + { TAS2770_INT_MASK_REG0, 0xfc }, + { TAS2770_INT_MASK_REG1, 0xb1 }, + { TAS2770_INT_CFG, 0x05 }, + { TAS2770_MISC_IRQ, 0x81 }, + { TAS2770_CLK_CGF, 0x0c }, + +}; + +static bool tas2770_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_PAGE: /* regmap implementation requires this */ + case TAS2770_SW_RST: /* always clears after write */ + case TAS2770_BO_PRV_REG0:/* has a self clearing bit */ + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0:/* Sticky interrupt flags */ + case TAS2770_LAT_INT_REG1:/* Sticky interrupt flags */ + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + return true; + } + return false; +} + +static bool tas2770_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS2770_LVE_INT_REG0: + case TAS2770_LVE_INT_REG1: + case TAS2770_LAT_INT_REG0: + case TAS2770_LAT_INT_REG1: + case TAS2770_VBAT_MSB: + case TAS2770_VBAT_LSB: + case TAS2770_TEMP_MSB: + case TAS2770_TEMP_LSB: + case TAS2770_TDM_CLK_DETC: + case TAS2770_REV_AND_GPID: + return false; + } + return true; +} + +static const struct regmap_range_cfg tas2770_regmap_ranges[] = { + { + .range_min = 0, + .range_max = 1 * 128, + .selector_reg = TAS2770_PAGE, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config tas2770_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = tas2770_writeable, + .volatile_reg = tas2770_volatile, + .reg_defaults = tas2770_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2770_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = tas2770_regmap_ranges, + .num_ranges = ARRAY_SIZE(tas2770_regmap_ranges), + .max_register = 1 * 128, +}; + +static irqreturn_t tas2770_irq_handler(int irq, void *dev_id) +{ + struct tas2770_priv *tas2770 = (struct tas2770_priv *)dev_id; + struct snd_soc_component *component = tas2770->component; + unsigned int device_status_1 = 0, device_status_2 = 0; + int result; + + if (tas2770->runtime_suspend) + goto end; + + if (tas2770->power_state == TAS2770_POWER_SHUTDOWN) + goto end; + + result = snd_soc_component_write(component, TAS2770_INT_MASK_REG0, + TAS2770_INT_MASK_REG0_DISABLE); + if (result) + goto reload; + result = snd_soc_component_write(component, TAS2770_INT_MASK_REG1, + TAS2770_INT_MASK_REG1_DISABLE); + if (result) + goto reload; + + result = tas2770_regmap_read(tas2770, + TAS2770_LAT_INT_REG0, &device_status_1); + if (!result && (device_status_1 & 0x3)) { + if (device_status_1 & + TAS2770_LAT_INT_REG0_OCE_FLG) { + tas2770->err_code |= ERROR_OVER_CURRENT; + dev_err(tas2770->dev, "SPK over current!\n"); + } else { + tas2770->err_code &= ~ERROR_OVER_CURRENT; + } + + if (device_status_1 & + TAS2770_LAT_INT_REG0_OTE_FLG) { + tas2770->err_code |= ERROR_DIE_OVERTEMP; + dev_err(tas2770->dev, "die over temperature!\n"); + } else { + tas2770->err_code &= ~ERROR_DIE_OVERTEMP; + } + } + + result = tas2770_regmap_read(tas2770, + TAS2770_LAT_INT_REG1, &device_status_2); + if (!result && (device_status_2 & 0x0f)) { + if (device_status_2 & + TAS2770_LAT_INT_REG1_VBA_TOV) { + tas2770->err_code |= ERROR_OVER_VOLTAGE; + dev_err(tas2770->dev, "SPK over voltage!\n"); + } else { + tas2770->err_code &= ~ERROR_OVER_VOLTAGE; + } + + if (device_status_2 & + TAS2770_LAT_INT_REG1_VBA_TUV) { + tas2770->err_code |= ERROR_UNDER_VOLTAGE; + dev_err(tas2770->dev, "SPK under voltage!\n"); + } else { + tas2770->err_code &= ~ERROR_UNDER_VOLTAGE; + } + + if (device_status_2 & + TAS2770_LAT_INT_REG1_BOUT_FLG) { + tas2770->err_code |= ERROR_BROWNOUT; + dev_err(tas2770->dev, "brownout!\n"); + } else { + tas2770->err_code &= ~ERROR_BROWNOUT; + } + } + + result = snd_soc_component_write(component, + TAS2770_INT_MASK_REG0, TAS2770_INT_REG0_DEFAULT); + if (result) + goto reload; + + result = snd_soc_component_write(component, + TAS2770_INT_MASK_REG1, TAS2770_INT_REG1_DEFAULT); + if (result) + goto reload; + + goto end; + +reload: + /* hardware reset and reload */ + tas2770_load_config(tas2770); + +end: + return IRQ_HANDLED; +} + +static int tas2770_parse_dt(struct device *dev, struct tas2770_priv *tas2770) +{ + int rc = 0; + + rc = fwnode_property_read_u32(dev->fwnode, "ti,asi-format", + &tas2770->asi_format); + if (rc) { + dev_err(tas2770->dev, "Looking up %s property failed %d\n", + "ti,asi-format", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &tas2770->i_sense_slot); + if (rc) { + dev_err(tas2770->dev, "Looking up %s property failed %d\n", + "ti,imon-slot-no", rc); + goto end; + } + + rc = fwnode_property_read_u32(dev->fwnode, "ti,vmon-slot-no", + &tas2770->v_sense_slot); + if (rc) { + dev_err(tas2770->dev, "Looking up %s property failed %d\n", + "ti,vmon-slot-no", rc); + goto end; + } + +end: + return rc; +} + +static int tas2770_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct tas2770_priv *tas2770; + int result; + + tas2770 = devm_kzalloc(&client->dev, + sizeof(struct tas2770_priv), GFP_KERNEL); + if (!tas2770) + return -ENOMEM; + tas2770->dev = &client->dev; + tas2770->irq_gpio = client->irq; + + i2c_set_clientdata(client, tas2770); + dev_set_drvdata(&client->dev, tas2770); + tas2770->power_state = TAS2770_POWER_SHUTDOWN; + + tas2770->regmap = devm_regmap_init_i2c(client, &tas2770_i2c_regmap); + if (IS_ERR(tas2770->regmap)) { + result = PTR_ERR(tas2770->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + result); + goto end; + } + + if (client->dev.of_node) { + result = tas2770_parse_dt(&client->dev, tas2770); + if (result) { + dev_err(tas2770->dev, "%s: Failed to parse devicetree\n", + __func__); + goto end; + } + } + + tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, + "reset-gpio", + GPIOD_OUT_HIGH); + if (IS_ERR(tas2770->reset_gpio)) { + if (PTR_ERR(tas2770->reset_gpio) == -EPROBE_DEFER) { + tas2770->reset_gpio = NULL; + return -EPROBE_DEFER; + } + } + + if (tas2770->irq_gpio > 0) { + result = devm_request_threaded_irq(tas2770->dev, + tas2770->irq_gpio, + NULL, tas2770_irq_handler, + IRQF_TRIGGER_FALLING | IRQF_ONESHOT, client->name, + tas2770); + if (result) { + dev_err(tas2770->dev, "Unable to request IRQ\n"); + return result; + } + } + + tas2770->channel_size = 0; + tas2770->slot_width = 0; + + tas2770_reset(tas2770); + + result = tas2770_register_codec(tas2770); + if (result) + dev_err(tas2770->dev, "Register codec failed.\n"); + +end: + return result; +} + +static int tas2770_i2c_remove(struct i2c_client *client) +{ + pm_runtime_disable(&client->dev); + return 0; +} + + +static const struct i2c_device_id tas2770_i2c_id[] = { + { "tas2770", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2770_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id tas2770_of_match[] = { + { .compatible = "ti,tas2770" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2770_of_match); +#endif + +static struct i2c_driver tas2770_i2c_driver = { + .driver = { + .name = "tas2770", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2770_of_match), + }, + .probe = tas2770_i2c_probe, + .remove = tas2770_i2c_remove, + .id_table = tas2770_i2c_id, +}; + +module_i2c_driver(tas2770_i2c_driver); + +MODULE_AUTHOR("Shi Fu <shifu0704(a)thundersoft.com>"); +MODULE_DESCRIPTION("TAS2770 I2C Smart Amplifier driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/tas2770.h b/sound/soc/codecs/tas2770.h new file mode 100644 index 0000000..bf59dc2 --- /dev/null +++ b/sound/soc/codecs/tas2770.h @@ -0,0 +1,166 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC TAS2770 codec driver + * + * Copyright (C) 2016-2017 Texas Instruments Incorporated - http://www.ti.com/ + */ +#ifndef __TAS2770__ +#define __TAS2770__ + +/* Book Control Register (available in page0 of each book) */ +#define TAS2770_BOOKCTL_PAGE 0 +#define TAS2770_BOOKCTL_REG 127 +#define TAS2770_REG(page, reg) ((page * 128) + reg) + /* Page */ +#define TAS2770_PAGE TAS2770_REG(0X0, 0x00) +#define TAS2770_PAGE_PAGE_MASK 255 + /* Software Reset */ +#define TAS2770_SW_RST TAS2770_REG(0X0, 0x01) +#define TAS2770_RST BIT(0) + /* Power Control */ +#define TAS2770_PWR_CTRL TAS2770_REG(0X0, 0x02) +#define TAS2770_PWR_CTRL_MASK 0x3 +#define TAS2770_PWR_CTRL_ACTIVE 0x0 +#define TAS2770_PWR_CTRL_MUTE BIT(0) +#define TAS2770_PWR_CTRL_SHUTDOWN 0x2 + /* Playback Configuration Reg0 */ +#define TAS2770_PLAY_CFG_REG0 TAS2770_REG(0X0, 0x03) + /* Playback Configuration Reg1 */ +#define TAS2770_PLAY_CFG_REG1 TAS2770_REG(0X0, 0x04) + /* Playback Configuration Reg2 */ +#define TAS2770_PLAY_CFG_REG2 TAS2770_REG(0X0, 0x05) +#define TAS2770_PLAY_CFG_REG2_VMAX 0xc9 + /* Misc Configuration Reg0 */ +#define TAS2770_MSC_CFG_REG0 TAS2770_REG(0X0, 0x07) + /* TDM Configuration Reg0 */ +#define TAS2770_TDM_CFG_REG0 TAS2770_REG(0X0, 0x0A) +#define TAS2770_TDM_CFG_REG0_SMP_MASK BIT(5) +#define TAS2770_TDM_CFG_REG0_SMP_48KHZ 0x0 +#define TAS2770_TDM_CFG_REG0_SMP_44_1KHZ BIT(5) +#define TAS2770_TDM_CFG_REG0_31_MASK 0xe +#define TAS2770_TDM_CFG_REG0_31_44_1_48KHZ 0x6 +#define TAS2770_TDM_CFG_REG0_31_88_2_96KHZ 0x8 +#define TAS2770_TDM_CFG_REG0_31_176_4_192KHZ 0xa + /* TDM Configuration Reg1 */ +#define TAS2770_TDM_CFG_REG1 TAS2770_REG(0X0, 0x0B) +#define TAS2770_TDM_CFG_REG1_MASK 0x3e +#define TAS2770_TDM_CFG_REG1_51_SHIFT 1 +#define TAS2770_TDM_CFG_REG1_RX_MASK BIT(0) +#define TAS2770_TDM_CFG_REG1_RX_RSING 0x0 +#define TAS2770_TDM_CFG_REG1_RX_FALING BIT(0) + /* TDM Configuration Reg2 */ +#define TAS2770_TDM_CFG_REG2 TAS2770_REG(0X0, 0x0C) +#define TAS2770_TDM_CFG_REG2_RXW_MASK 0xc +#define TAS2770_TDM_CFG_REG2_RXW_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXW_24BITS 0x8 +#define TAS2770_TDM_CFG_REG2_RXW_32BITS 0xc +#define TAS2770_TDM_CFG_REG2_RXS_MASK 0x3 +#define TAS2770_TDM_CFG_REG2_RXS_16BITS 0x0 +#define TAS2770_TDM_CFG_REG2_RXS_24BITS BIT(0) +#define TAS2770_TDM_CFG_REG2_RXS_32BITS 0x2 + /* TDM Configuration Reg3 */ +#define TAS2770_TDM_CFG_REG3 TAS2770_REG(0X0, 0x0D) +#define TAS2770_TDM_CFG_REG3_RXS_MASK 0xf0 +#define TAS2770_TDM_CFG_REG3_RXS_SHIFT 0x4 +#define TAS2770_TDM_CFG_REG3_30_MASK 0xf +#define TAS2770_TDM_CFG_REG3_30_SHIFT 0 + /* TDM Configuration Reg5 */ +#define TAS2770_TDM_CFG_REG5 TAS2770_REG(0X0, 0x0F) +#define TAS2770_TDM_CFG_REG5_VSNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG5_VSNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG5_50_MASK 0x3f + /* TDM Configuration Reg6 */ +#define TAS2770_TDM_CFG_REG6 TAS2770_REG(0X0, 0x10) +#define TAS2770_TDM_CFG_REG6_ISNS_MASK BIT(6) +#define TAS2770_TDM_CFG_REG6_ISNS_ENABLE BIT(6) +#define TAS2770_TDM_CFG_REG6_50_MASK 0x3f + /* Brown Out Prevention Reg0 */ +#define TAS2770_BO_PRV_REG0 TAS2770_REG(0X0, 0x1B) + /* Interrupt MASK Reg0 */ +#define TAS2770_INT_MASK_REG0 TAS2770_REG(0X0, 0x20) +#define TAS2770_INT_REG0_DEFAULT 0xfc +#define TAS2770_INT_MASK_REG0_DISABLE 0xff + /* Interrupt MASK Reg1 */ +#define TAS2770_INT_MASK_REG1 TAS2770_REG(0X0, 0x21) +#define TAS2770_INT_REG1_DEFAULT 0xb1 +#define TAS2770_INT_MASK_REG1_DISABLE 0xff + /* Live-Interrupt Reg0 */ +#define TAS2770_LVE_INT_REG0 TAS2770_REG(0X0, 0x22) + /* Live-Interrupt Reg1 */ +#define TAS2770_LVE_INT_REG1 TAS2770_REG(0X0, 0x23) + /* Latched-Interrupt Reg0 */ +#define TAS2770_LAT_INT_REG0 TAS2770_REG(0X0, 0x24) +#define TAS2770_LAT_INT_REG0_OCE_FLG BIT(1) +#define TAS2770_LAT_INT_REG0_OTE_FLG BIT(0) + /* Latched-Interrupt Reg1 */ +#define TAS2770_LAT_INT_REG1 TAS2770_REG(0X0, 0x25) +#define TAS2770_LAT_INT_REG1_VBA_TOV BIT(3) +#define TAS2770_LAT_INT_REG1_VBA_TUV BIT(2) +#define TAS2770_LAT_INT_REG1_BOUT_FLG BIT(1) + /* VBAT MSB */ +#define TAS2770_VBAT_MSB TAS2770_REG(0X0, 0x27) + /* VBAT LSB */ +#define TAS2770_VBAT_LSB TAS2770_REG(0X0, 0x28) + /* TEMP MSB */ +#define TAS2770_TEMP_MSB TAS2770_REG(0X0, 0x29) + /* TEMP LSB */ +#define TAS2770_TEMP_LSB TAS2770_REG(0X0, 0x2A) + /* Interrupt Configuration */ +#define TAS2770_INT_CFG TAS2770_REG(0X0, 0x30) + /* Misc IRQ */ +#define TAS2770_MISC_IRQ TAS2770_REG(0X0, 0x32) + /* Clock Configuration */ +#define TAS2770_CLK_CGF TAS2770_REG(0X0, 0x3C) + /* TDM Clock detection monitor */ +#define TAS2770_TDM_CLK_DETC TAS2770_REG(0X0, 0x77) + /* Revision and PG ID */ +#define TAS2770_REV_AND_GPID TAS2770_REG(0X0, 0x7D) + +#define TAS2770_POWER_ACTIVE 0 +#define TAS2770_POWER_MUTE 1 +#define TAS2770_POWER_SHUTDOWN 2 +#define ERROR_OVER_CURRENT 0x0000001 +#define ERROR_DIE_OVERTEMP 0x0000002 +#define ERROR_OVER_VOLTAGE 0x0000004 +#define ERROR_UNDER_VOLTAGE 0x0000008 +#define ERROR_BROWNOUT 0x0000010 +#define ERROR_CLASSD_PWR 0x0000020 +#define TAS2770_SLOT_16BIT 16 +#define TAS2770_SLOT_32BIT 32 +#define TAS2770_I2C_RETRY_COUNT 3 + +struct tas2770_register { + int book; + int page; + int reg; +}; + +struct tas2770_dai_cfg { + unsigned int dai_fmt; + unsigned int tdm_delay; +}; + +struct tas2770_priv { + struct device *dev; + struct regmap *regmap; + struct snd_soc_codec *codec; + struct snd_soc_component *component; + struct mutex dev_lock; + struct hrtimer mtimer; + int power_state; + int asi_format; + struct gpio_desc *reset_gpio; + int irq_gpio; + bool irq_enable; + int sampling_rate; + int frame_size; + int channel_size; + int slot_width; + int v_sense_slot; + int i_sense_slot; + bool runtime_suspend; + unsigned int err_code; + struct mutex codec_lock; +}; + +#endif /* __TAS2770__ */ -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: dmaengine: Replace strncpy() with strscpy_pad() for pcm->name
by Peter Ujfalusi 11 Sep '19

11 Sep '19

While it is safe to use strncpy in this case, the advice is to move to strscpy or strscpy_pad. Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index d93db2c2b527..5552c66ca642 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -308,9 +308,9 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; if (rtd->pcm->streams[i].pcm->name[0] == '\0') { - strncpy(rtd->pcm->streams[i].pcm->name, - rtd->pcm->streams[i].pcm->id, - sizeof(rtd->pcm->streams[i].pcm->name)); + strscpy_pad(rtd->pcm->streams[i].pcm->name, + rtd->pcm->streams[i].pcm->id, + sizeof(rtd->pcm->streams[i].pcm->name)); } } -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: wcd9335: remove redundant use of ret variable
by Saiyam Doshi 11 Sep '19

11 Sep '19

All these functions declares and initializes variable ret with '0' and without modifying 'ret' variable, it is returned. This patch removes this redundancy and returns '0' directly. Signed-off-by: Saiyam Doshi <saiyamdoshi.in(a)gmail.com> --- sound/soc/codecs/wcd9335.c | 24 ++++++++---------------- 1 file changed, 8 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 03f8a94bba2f..f318403133e9 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -3022,7 +3022,6 @@ static int wcd9335_codec_enable_slim(struct snd_soc_dapm_widget *w, struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = snd_soc_component_get_drvdata(comp); struct wcd_slim_codec_dai_data *dai = &wcd->dai[w->shift]; - int ret = 0; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -3034,7 +3033,7 @@ static int wcd9335_codec_enable_slim(struct snd_soc_dapm_widget *w, break; } - return ret; + return 0; } static int wcd9335_codec_enable_mix_path(struct snd_soc_dapm_widget *w, @@ -3539,7 +3538,6 @@ static int wcd9335_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; u8 dem_inp; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3579,7 +3577,7 @@ static int wcd9335_codec_hphl_dac_event(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_lineout_dac_event(struct snd_soc_dapm_widget *w, @@ -3607,7 +3605,6 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w, { struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3621,7 +3618,7 @@ static int wcd9335_codec_ear_dac_event(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static void wcd9335_codec_hph_post_pa_config(struct wcd9335_codec *wcd, @@ -3692,7 +3689,6 @@ static int wcd9335_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; u8 dem_inp; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3731,7 +3727,7 @@ static int wcd9335_codec_hphr_dac_event(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, @@ -3741,7 +3737,6 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3780,7 +3775,7 @@ static int wcd9335_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w, @@ -3789,7 +3784,6 @@ static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w, { struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); int vol_reg = 0, mix_vol_reg = 0; - int ret = 0; if (w->reg == WCD9335_ANA_LO_1_2) { if (w->shift == 7) { @@ -3837,7 +3831,7 @@ static int wcd9335_codec_enable_lineout_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static void wcd9335_codec_init_flyback(struct snd_soc_component *component) @@ -3892,7 +3886,6 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); struct wcd9335_codec *wcd = dev_get_drvdata(comp->dev); int hph_mode = wcd->hph_mode; - int ret = 0; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -3930,14 +3923,13 @@ static int wcd9335_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kc, int event) { struct snd_soc_component *comp = snd_soc_dapm_to_component(w->dapm); - int ret = 0; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -3967,7 +3959,7 @@ static int wcd9335_codec_enable_ear_pa(struct snd_soc_dapm_widget *w, break; }; - return ret; + return 0; } static irqreturn_t wcd9335_slimbus_irq(int irq, void *data) -- 2.20.1

2 1

[alsa-devel] [PATCH v2] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 11 Sep '19

11 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: fsl_mqs: add DT binding documentation
by Shengjiu Wang 11 Sep '19

11 Sep '19

Add the DT binding documentation for NXP MQS driver Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../devicetree/bindings/sound/fsl,mqs.txt | 20 +++++++++++++++++++ 1 file changed, 20 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt new file mode 100644 index 000000000000..a1dbe181204a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -0,0 +1,20 @@ +fsl,mqs audio CODEC + +Required properties: + + - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + - clocks : A list of phandles + clock-specifiers, one for each entry in + clock-names + - clock-names : Must contain "mclk" + - gpr : The gpr node. + +Example: + +mqs: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + status = "disabled"; +}; -- 2.21.0

3 4

[alsa-devel] [PATCH] tas2770: add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 11 Sep '19

11 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- Documentation/devicetree/bindings/tas2770.txt | 38 +++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) create mode 100644 Documentation/devicetree/bindings/tas2770.txt diff --git a/Documentation/devicetree/bindings/tas2770.txt b/Documentation/devicetree/bindings/tas2770.txt new file mode 100644 index 0000000..f70b310 --- /dev/null +++ b/Documentation/devicetree/bindings/tas2770.txt @@ -0,0 +1,38 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,left-slot: - Sets TDM RX left time slots. + - ti,right-slot: - Sets TDM RX right time slots. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + + - reset-gpio: Reset GPIO number of left device. + - irq-gpio: IRQ GPIO number of left device. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + reset-gpio = <&gpio15 1 GPIO_ACTIVE_LOW>; + irq-gpio = <&gpio16 1 GPIO_ACTIVE_LOW>; + ti,asi-format = <0>; + ti,left-slot = <0>; + ti,right-slot = <1>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

3 7

[alsa-devel] [PATCH] ASoC: dmaengine: Make the pcm->name equal to pcm->id if the name is not set
by Peter Ujfalusi 11 Sep '19

11 Sep '19

Some tools use the snd_pcm_info_get_name() to try to identify PCMs or for other purposes. Currently it is left empty with the dmaengine-pcm, in this case copy the pcm->id string as pcm->name. For example IGT is using this to find the HDMI PCM for testing audio on it. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Reported-by: Arthur She <arthur.she(a)linaro.org> --- Hi, this was actually reported for 4.14 kernel with omap-pcm (replaced by sdma-pcm in v4.18), since then we only use the generic dmaengine PCM but the same issue applies today. Regards, Peter sound/soc/soc-generic-dmaengine-pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 748f5f641002..d93db2c2b527 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -306,6 +306,12 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!dmaengine_pcm_can_report_residue(dev, pcm->chan[i])) pcm->flags |= SND_DMAENGINE_PCM_FLAG_NO_RESIDUE; + + if (rtd->pcm->streams[i].pcm->name[0] == '\0') { + strncpy(rtd->pcm->streams[i].pcm->name, + rtd->pcm->streams[i].pcm->id, + sizeof(rtd->pcm->streams[i].pcm->name)); + } } return 0; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 7

[alsa-devel] [PATCH 00/13] ASoC: soc-core cleanup - step 3
by Kuninori Morimoto 11 Sep '19

11 Sep '19

Hi Mark These are soc-core.c cleanup (maybe) step 3. These try to clenaup around kmalloc/kfree related code. Basically these do nothing from technical point of view. Kuninori Morimoto (13): ASoC: soc-core: move soc_free_pcm_runtime() ASoC: soc-core: merge soc_add_pcm_runtime() into soc_new_pcm_runtime() ASoC: soc-core: call list_del(&rtd->list) at soc_free_pcm_runtime() ASoC: soc-core: create rtd->codec_dais first ASoC: soc-core: merge soc_new_pcm_runtime() and soc_rtd_init() ASoC: soc-core: merge soc_free_pcm_runtime() and soc_rtd_free() ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order ASoC: soc-core: remove snd_soc_rtdcom_del_all() ASoC: soc-core: use devm_kzalloc() for rtd ASoC: soc-core: remove soc_remove_dai_links() ASoC: soc-core: add soc_setup_card_name() ASoC: soc-core: use devm_xxx for component related resource ASoC: soc-core: setup card->rtd_num at snd_soc_instantiate_card() include/sound/soc.h | 1 - sound/soc/soc-core.c | 276 +++++++++++++++++++++++++-------------------------- 2 files changed, 138 insertions(+), 139 deletions(-) -- 2.7.4

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[alsa-devel] Measuring Device Throughput - MMAP vs. RW
by Pavel Hofman 10 Sep '19

10 Sep '19

Hi, Gathering data for USB async gadget feedback ( https://mailman.alsa-project.org/pipermail/alsa-devel/2019-August/154819.ht… ) requires measuring average data rate to/from a device. For testing I added measuring/averaging code to methods snd_pcm_hw_writei, snd_pcm_hw_readi, and snd_pcm_hw_mmap_commit in https://github.com/alsa-project/alsa-lib/blob/master/src/pcm/pcm_hw.c . In all cases I take the parameter "size" of all the three methods as the number of samples written/read. RW: The averaging works very well for R/W access (hw:X) for playback and capture, I get same values on the same device (playback/capture timed by one clock). MMAP: When mmap is used (e.g. plughw:X + sample format conversion), I get very stable results for MMAP playback on all soundcards I have tested. But for capture I get identical results to playback for one soundcard (PCI Envy24), while the averaged rate calculated from "size" in snd_pcm_hw_mmap_commit fluctuates for capture on Intel HDA. The averaging is simply summing "size" in every call of the method and dividing by passed nanosecs every approx 10 seconds: Working OK (Infrasonic Quartet, and all soundcards for read/write mode): Playback MMAP Time: 19440.900391, averaged samplerate: 48000.511829 MMAP Time: 19451.140625, averaged samplerate: 48000.360218 MMAP Time: 19461.380859, averaged samplerate: 48000.176846 MMAP Time: 19471.619141, averaged samplerate: 48000.664570 MMAP Time: 19481.861328, averaged samplerate: 48000.199970 MMAP Time: 19492.101562, averaged samplerate: 48000.589854 MMAP Time: 19502.341797, averaged samplerate: 48000.288977 MMAP Time: 19512.580078, averaged samplerate: 48000.516104 MMAP Time: 19522.820312, averaged samplerate: 48000.225282 Capture: MMAP Time: 19454.578125, averaged samplerate: 48000.430766 MMAP Time: 19464.818359, averaged samplerate: 48000.397611 MMAP Time: 19475.056641, averaged samplerate: 48000.400049 MMAP Time: 19485.296875, averaged samplerate: 48000.293275 MMAP Time: 19495.539062, averaged samplerate: 48000.549048 MMAP Time: 19505.779297, averaged samplerate: 48000.445214 MMAP Time: 19516.017578, averaged samplerate: 48000.417299 MMAP Time: 19526.257812, averaged samplerate: 48000.376264 MMAP Time: 19536.498047, averaged samplerate: 48000.422957 MMAP Time: 19546.738281, averaged samplerate: 48000.538619 MMAP Time: 19556.976562, averaged samplerate: 48000.241647 MMAP Time: 19567.216797, averaged samplerate: 48000.447243 Fluctuating Capture (Intel HDA): Playback - OK: MMAP Time: 19574.498047, averaged samplerate: 47998.971874 MMAP Time: 19584.568359, averaged samplerate: 47998.871734 MMAP Time: 19594.636719, averaged samplerate: 47998.929850 MMAP Time: 19604.707031, averaged samplerate: 47998.972555 MMAP Time: 19614.775391, averaged samplerate: 47998.925531 MMAP Time: 19624.845703, averaged samplerate: 47998.893351 MMAP Time: 19634.916016, averaged samplerate: 47998.949489 MMAP Time: 19644.986328, averaged samplerate: 47998.593678 MMAP Time: 19655.056641, averaged samplerate: 47999.230857 MMAP Time: 19665.125000, averaged samplerate: 47998.777034 MMAP Time: 19675.195312, averaged samplerate: 47999.177460 MMAP Time: 19685.263672, averaged samplerate: 47998.798503 Capture - fluctuating, clearly some samples are not counted: MMAP Time: 19539.853516, averaged samplerate: 47191.039837 MMAP Time: 19549.923828, averaged samplerate: 46369.921890 MMAP Time: 19559.994141, averaged samplerate: 47998.831508 MMAP Time: 19570.148438, averaged samplerate: 47190.998307 MMAP Time: 19580.302734, averaged samplerate: 47191.275432 MMAP Time: 19590.373047, averaged samplerate: 47998.894343 MMAP Time: 19600.443359, averaged samplerate: 47998.906779 MMAP Time: 19610.511719, averaged samplerate: 47999.174771 MMAP Time: 19620.582031, averaged samplerate: 47998.747971 MMAP Time: 19630.650391, averaged samplerate: 47998.881964 MMAP Time: 19640.722656, averaged samplerate: 47998.999974 MMAP Time: 19650.791016, averaged samplerate: 47998.769240 I suspect the size parameter of snd_pcm_hw_mmap_commit is not the actual amount of samples read during the MMAP capture. Please how to calculate the number of samples read from the variables available in that method? Thanks a lot. Pavel.

2 7

[alsa-devel] [PATCH 0/2] ALSA: firewire-tascam: fix to handle error code to read status of sampling clock
by Takashi Sakamoto 10 Sep '19

10 Sep '19

Hi, ALSA firewire-tascam driver has a bug not to check return value from a call of snd_tscm_stream_get_clock() since its first commit. This patchset fixes the bug. In models of TASCAM FIreWire series, the status of sampling clock is represented in 2 bytes in MSB of a certain register. These bits are zero in intermediate state after configuring the sampling clock. This behaviour sometimes takes ALSA PCM applications to receive error when starting both playback/capture PCM substreams because the operation to start one PCM substream can change the sampling clock. In this case, next operation for another PCM substream possible fails. This patchset adds check and retry for the above case. Although the change is a bit intrusive for stable releases, it's required to avoid superficial regression due to the additional error check. Takashi Sakamoto (2): ALSA: firewire-tascam: handle error code when getting current source of clock ALSA: firewire-tascam: check intermediate state of clock status and retry sound/firewire/tascam/tascam-pcm.c | 3 ++ sound/firewire/tascam/tascam-stream.c | 42 ++++++++++++++++++--------- 2 files changed, 31 insertions(+), 14 deletions(-) -- 2.20.1

2 4

[alsa-devel] [GIT PULL] ASoC updates v5.4
by Mark Brown 10 Sep '19

10 Sep '19

The following changes since commit f74c2bb98776e2de508f4d607cd519873065118e: Linux 5.3-rc8 (2019-09-08 13:33:15 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git asoc-v5.4 for you to fetch changes up to bb831786117519fc16dfd3eaa7b84e4f6bbb8d99: Merge branch 'asoc-5.4' into asoc-next (2019-09-09 14:55:20 +0100) ---------------------------------------------------------------- ASoC: Updates for v5.4 Quite a big update this time around, particularly in the core where we've had a lot of cleanups from Morimoto-san - there's not much functional change but quite a bit of modernization going on. We've also seen a lot of driver work, a lot of it cleanups but also some particular drivers. - Lots and lots of cleanups from Morimoto-san and Yue Haibing. - Lots of cleanups and enhancements to the Freescale, sunxi dnd Intel rivers. - Initial Sound Open Firmware suppot for i.MX8. - Removal of w90x900 and nuc900 drivers as the platforms are being removed. - New support for Cirrus Logic CS47L15 and CS47L92, Freescale i.MX 7ULP and 8MQ, Meson G12A and NXP UDA1334 ---------------------------------------------------------------- Amadeusz Sławiński (9): ASoC: Intel: Skylake: Combine snd_soc_skl_ipc and snd_soc_skl ASoC: Intel: Skylake: Do not disable FW notifications ASoC: Intel: Skylake: Remove static table index when parsing topology ASoC: Intel: Skylake: Use correct function to access iomem space ASoC: Intel: Fix use of potentially uninitialized variable ASoC: dapm: Expose snd_soc_dapm_new_control_unlocked properly ASoC: Intel: NHLT: Fix debug print format ASoC: Intel: Skylake: Print module type instead of id ASoC: Intel: Skylake: Release topology when we are done with it Andra Danciu (2): dt-bindings: sound: Add bindings for UDA1334 codec ASoC: codecs: Add uda1334 codec driver Andy Shevchenko (1): ASoC: Intel: Skylake: Print constant literals from format specifier Arnd Bergmann (1): ASoC: remove w90x900/nuc900 platform drivers Bard liao (1): ASoC: SOF: rename SOUNDWIRE to ALH Ben Zhang (1): ASoC: rt5677: keep analog power register at SND_SOC_BIAS_OFF Bjorn Andersson (1): ASoC: qcom: common: Include link-name in error messages Cezary Rojewski (13): ASoC: Intel: Skylake: Merge skl_sst and skl into skl_dev struct ASoC: Intel: Skylake: Remove MCPS available check ASoC: Intel: Skylake: Remove memory available check ASoC: Intel: Skylake: Make MCPS and CPS params obsolete ASoC: Intel: Skylake: Cleanup skl_module_cfg declaration ASoC: Intel: Update request-reply IPC model ASoC: Intel: Skylake: Limit large_config_get to single frame ASoC: Intel: Skylake: large_config_get overhaul ASoC: Intel: Baytrail: Fix implicit fallthrough warning ASoC: Intel: Haswell: Adjust machine device private context ASoC: Intel: haswell: Simplify device probe ASoC: Intel: bdw-rt5677: Simplify device probe ASoC: Intel: broadwell: Simplify device probe Charles Keepax (3): ASoC: cs47l35: Use define for DSPCLK enable bit ASoC: madera: Read device tree configuration ASoC: wm_adsp: Allow bus error handler to be called directly Cheng-Yi Chiang (2): ASoC: rockchip: rockchip_max98090: Set period size to 240 ASoC: hdmi-codec: Add an op to set callback function for plug event Christophe JAILLET (4): ASoC: pcm3168a: Fix a typo in the name of a constant ASoC: rt5665: Fix a typo in the name of a function ASoC: wm8955: Fix a typo in 'wm8995_pll_factors()' function name ASoC: cs4271: Fix a typo in the CS4171_NR_RATIOS Codrin Ciubotariu (4): ASoC: codecs: ad193x: Use regmap_multi_reg_write() when initializing ASoC: mchp-i2s-mcc: Fix unprepare of GCLK ASoC: mchp-i2s-mcc: Wait for RX/TX RDY only if controller is running ASoC: mchp-i2s-mcc: Fix simultaneous capture and playback in master mode Colin Ian King (6): ASoC: codec2codec: fix missing return of error return code ASoC: zx-tdm: remove redundant assignment to ts_width on error return path ASoC: codecs: ad193x: make two arrays static const, makes object smaller ASoC: rt1011: make array pd static const, makes object smaller ASoC: rt1305: make array pd static const, makes object smaller ASoC: rt1308: make array pd static const, makes object smaller Curtis Malainey (3): ASoC: rt5677: remove superfluous set ASoC: rt5677: Revert remove superfluous set ASoC: rt5677: Remove magic number register writes Dan Carpenter (1): ASoC: SOF: imx8: Fix an is IS_ERR() vs NULL check Daniel Baluta (20): ASoC: SOF: Introduce snd_sof_dsp_get_bar_index ops ASoC: SOF: loader: Use the BAR provided by FW ASoC: SOF: loader: Don't ignore SRAM block types ASoC: dt-bindings: Introduce compatible strings for 7ULP and 8MQ ASoC: fsl_sai: Add registers definition for multiple datalines ASoC: fsl_sai: Update Tx/Rx channel enable mask ASoC: fsl_sai: Add support for SAI new version ASoC: fsl_sai: Add support for imx7ulp/imx8mq ASoC: SOF: Remove call to snd_sof_dsp_mailbox_init ASoC: SOF: Intel: bdw: Use generic function for fw ready / mem windows creation ASoC: SOF: Intel: hda: Use generic function for fw ready / mem windows creation ASoC: SOF: Introduce snd_sof_dsp_get_mailbox_offset ASoC: SOF: Introduce snd_sof_dsp_get_window_offset ASoC: SOF: Intel: byt: Refactor fw ready / mem windows creation ASoC: SOF: topology: Add dummy support for i.MX8 DAIs ASoC: fsl_sai: Add support for imx8qm ASoC: dt-bindings: Introduce compatible string for imx8qm dt-bindings: dsp: fsl: Add DSP core binding support ASoC: SOF: Add OF DSP device support ASoC: SOF: imx: Add i.MX8 HW support Daniel Stuart (2): ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk ASoC: intel: cht_bsw_max98090_ti: Add all Chromebooks that need pmc_plt_clk_0 quirk Enric Balletbo i Serra (1): ASoC: rockchip-max98090: Remove MICBIAS as supply of input pin IN34 Fred Oh (1): ASoC: SOF: remove unused state variable in suspend function Greg Kroah-Hartman (3): ASoC: Intel: SoC: skylake: no need to check return value of debugfs_create functions ASoC: core: no need to check return value of debugfs_create functions ASoC: SOF: no need to check return value of debugfs_create functions Guennadi Liakhovetski (1): ASoC: SOF: Intel: hda: add a parameter to disable MSI Gustavo A. R. Silva (1): ASoC: ti: Mark expected switch fall-throughs Hariprasad Kelam (3): sound: soc: codecs: mt6358: change return type of mt6358_codec_init_reg ASoC: bcm: cygnus-pcm: Unneeded variable: "ret". ASoC: wcd9335: add irqflag IRQF_ONESHOT flag Janusz Jankowski (1): ASoC: SOF: Intel: ssp: BCLK delay parameter Janusz Krzysztofik (1): ASoC: ams-delta: Take control over audio mute GPIO pins Jaroslav Kysela (1): ASoC: SOF: Makefile - fix the top-level kernel module names (add snd- prefix) Jaska Uimonen (2): ASoC: SOF: topology: use set_get_data in process load ASoC: SOF: topology: fix get control data return type and arguments Jerome Brunet (17): ASoC: codec2codec: run callbacks in order ASoC: codec2codec: name link using stream direction ASoC: codec2codec: deal with params when necessary ASoC: meson: g12a-tohdmitx: override codec2codec params ASoC: create pcm for codec2codec links as well ASoC: codec2codec: remove ephemeral variables ASoC: codec2codec: fill some of the runtime stream parameters ASoC: meson: axg-tdm-formatter: free reset on device removal ASoC: meson: g12a-tohdmitx: require regmap mmio ASoC: meson: add sm1 compatibles ASoC: meson: add reset binding ASoC: meson: axg-frddr: expose all 8 outputs ASoC: meson: axg-toddr: expose all 8 inputs ASoC: meson: tdmin: expose all 16 inputs ASoC: meson: axg-frddr: add sm1 support ASoC: meson: axg-toddr: add sm1 support ASoC: meson: tdmout: add sm1 support Jiaxin Yu (5): ASoC: mediatek: mt6358: add delay after dmic clock on ASoC: mediatek: mt8183: switch tdm pins gpio function when playback on or off ASoC: mediatek: mt8183: fix tdm out data align issue ASoC: mediatek: mt8183: tdm hw support tdm out and 8ch i2s out ASoC: mediatek: mt8183: fix tdm out data is valid on rising edge Joe Perches (1): ASoC: wcd9335: Fix misuse of GENMASK macro Kai Vehmanen (5): ASoC: SOF: reset DMA state in prepare ASoC: SOF: use common code to send PCM_FREE IPC ASoC: SOF: ipc: use timeout configured at probe ASoC: SOF: core: increase default IPC timeouts ASoC: SOF: Intel: hda: reset link DMA state in prepare Katsuhiro Suzuki (8): ASoC: es8316: fix headphone mixer volume table ASoC: es8316: fix inverted L/R of headphone mixer volume ASoC: es8316: add DT-bindings ASoC: es8316: add clock control of MCLK ASoC: es8316: fix redundant codes of clock ASoC: es8316: support fixed and variable both clock rates ASoC: rockchip: ignore 0Hz sysclk SoC: simple-card-utils: set 0Hz to sysclk when shutdown Keyon Jie (5): ASoC: SOF: Intel: hda: correct ROM state mask ASoC: hdac_hda: fix page fault issue by removing race ASoC: Intel: sof-rt5682: add dmic dapm widget to support dmic PCM ASoC: Intel: skl-hda-dsp-generic: add dependency to dmic driver ASoC: Intel: skl-hda-dsp-generic: add dmic dapm widget and route Kunihiko Hayashi (1): ASoC: uniphier: Fix double reset assersion when transitioning to suspend state Kuninori Morimoto (101): ASoC: add soc-dai.c ASoC: soc-dai: mv soc_dai_hw_params() to soc-dai ASoC: soc-dai: add snd_soc_dai_hw_free() ASoC: soc-dai: add snd_soc_dai_startup() ASoC: soc-dai: add snd_soc_dai_shutdown() ASoC: soc-dai: add snd_soc_dai_prepare() ASoC: soc-dai: add snd_soc_dai_trigger() ASoC: soc-dai: add snd_soc_dai_bespoke_trigger() ASoC: soc-dai: add snd_soc_dai_delay() ASoC: soc-dai: add snd_soc_dai_suspend() ASoC: soc-dai: add snd_soc_dai_resume() ASoC: soc-dai: add snd_soc_dai_probe() ASoC: soc-dai: add snd_soc_dai_remove() ASoC: soc-dai: add snd_soc_dai_compress_new() ASoC: soc-core: use existing snd_soc_dai_digital_mute() ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c ASoC: soc-pcm: remove soc_rtdcom_ack() ASoC: soc-pcm: remove soc_rtdcom_copy_kernel() ASoC: soc-pcm: remove soc_fill_silence() ASoC: add soc-component.c ASoC: soc-component: add snd_soc_component_get/put() ASoC: soc-component: add snd_soc_component_open() ASoC: soc-component: add snd_soc_component_close() ASoC: soc-component: add snd_soc_component_prepare() ASoC: soc-component: add snd_soc_component_hw_params() ASoC: soc-component: add snd_soc_component_hw_free() ASoC: soc-component: add snd_soc_component_trigger() ASoC: soc-component: add snd_soc_component_suspend() ASoC: soc-component: add snd_soc_component_resume() ASoC: soc-component: add snd_soc_component_is_suspended() ASoC: soc-component: add snd_soc_component_probe() ASoC: soc-component: add snd_soc_component_remove() ASoC: soc-component: add snd_soc_component_of_xlate_dai_id() ASoC: soc-component: add snd_soc_component_of_xlate_dai_name() ASoC: soc-component: move snd_soc_component_seq_notifier() ASoC: soc-component: move snd_soc_component_stream_event() ASoC: soc-component: move snd_soc_component_set_bias_level() ASoC: soc-component: add snd_soc_pcm_component_pointer() ASoC: soc-component: add snd_soc_pcm_component_ioctrl() ASoC: soc-component: add snd_soc_pcm_component_copy_user() ASoC: soc-component: add snd_soc_pcm_component_page() ASoC: soc-component: add snd_soc_pcm_component_mmap() ASoC: soc-component: add snd_soc_pcm_component_pcm_new() ASoC: soc-component: add snd_soc_pcm_component_pcm_free() ASoC: rsnd: don't call clk_get_rate() under atomic context ASoC: soc-core: set component->debugfs_root NULL ASoC: soc-core: add comment for for_each_xxx ASoC: soc-core: check return value of snd_soc_add_dai_link() ASoC: soc-core: don't use for_each_card_links_safe() at snd_soc_find_dai_link() ASoC: soc-core: reuse rtdcom at snd_soc_rtdcom_add() ASoC: soc-core: tidyup for snd_soc_dapm_new_controls() ASoC: soc-core: tidyup for snd_soc_add_component_controls() ASoC: soc-core: tidyup for snd_soc_dapm_add_routes() ASoC: soc-core: tidyup for snd_soc_add_card_controls() ASoC: soc-core: call snd_soc_dapm_debugfs_init() at soc_init_card_debugfs() ASoC: soc-core: remove unneeded list_empty() check for snd_soc_try_rebind_card() ASoC: soc-core: tidyup for card->deferred_resume_work ASoC: soc-core: define soc_dpcm_debugfs_add() for non CONFIG_DEBUG_FS ASoC: soc-core: dai_link check under soc_dpcm_debugfs_add() ASoC: soc-core: support snd_soc_dai_link_component for aux_dev ASoC: simple-card: use snd_soc_dai_link_component for aux_dev ASoC: intel: cht_bsw_max98090_ti: use snd_soc_dai_link_component for aux_dev ASoC: mediatek: mt8183-da7219-max98357: use snd_soc_dai_link_component for aux_dev ASoC: meson: axg-card: use snd_soc_dai_link_component for aux_dev ASoC: rockchip: rockchip_max98090: use snd_soc_dai_link_component for aux_dev ASoC: samsung: neo1973_wm8753: use snd_soc_dai_link_component for aux_dev ASoC: samsung: speyside: use snd_soc_dai_link_component for aux_dev ASoC: samsung: tm2_wm5110: use snd_soc_dai_link_component for aux_dev ASoC: sunxi: sun4i-codec: use snd_soc_dai_link_component for aux_dev ASoC: ti: rx51: use snd_soc_dai_link_component for aux_dev ASoC: soc-core: remove legacy style of aux_dev ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: use snd_soc_dai_link_component for aux_dev ASoC: soc-core: add for_each_xxx macro for aux_dev ASoC: soc-core: merge snd_soc_initialize_card_lists() ASoC: soc-core: remove unneeded dai_link check from snd_soc_remove_dai_link() ASoC: soc-core: use device_register() ASoC: soc-core: add NOTE to snd_soc_rtdcom_lookup() ASoC: soc-core: soc_cleanup_card_resources() become void ASoC: soc-dai: use bit field for bus_control ASoC: soc-topology: use for_each_component_dais() at remove_dai() ASoC: soc-core: initialize component list ASoC: soc-core: initialize list at one place ASoC: soc-core: rename soc_post_component_init() to soc_rtd_init() ASoC: soc-core: add soc_rtd_free() ASoC: soc-core: move soc_probe_component() position ASoC: soc-core: dapm related setup at one place ASoC: soc-core: add snd_soc_dapm_init() ASoC: soc-core: move soc_probe_link_components() position ASoC: simple-card: indicate rebind issue ASoC: audio-graph: indicate rebind issue ASoC: soc-core: add comment to jack at soc_remove_component() ASoC: soc-core: self contained soc_probe_link_components() ASoC: soc-core: self contained soc_remove_link_components() ASoC: soc-core: self contained soc_remove_link_dais() ASoC: soc-core: move soc_probe_dai() next to soc_remove_dai() ASoC: soc-core: add new soc_link_init() ASoC: soc-core: self contained soc_probe_link_dais() ASoC: soc-core: move soc_probe_link_dais() next to soc_remove_link_dais() ASoC: soc-core: self contained soc_bind_aux_dev() ASoC: soc-core: add soc_unbind_aux_dev() ASoC: soc-core: self contained soc_unbind_aux_dev() Libin Yang (1): ASoC: intel: skl_hda_dsp_common: create HDMI jack kctl Lucas Stach (4): ASoC: fsl_sai: add of_match data ASoC: fsl_sai: derive TX FIFO watermark from FIFO depth ASoC: fsl_sai: mark regmap as fast_io ASoC: tlv320aic31xx: suppress error message for EPROBE_DEFER Mac Chiang (1): ASoC: Intel: boards: Add Cometlake machine driver support Mark Brown (8): Merge branch 'asoc-5.3' into asoc-5.4 Merge branch 'topic/hda-dmic' of https://git.kernel.org/.../tiwai/sound into asoc-5.4 ASoC: sof: Fix warning when IPC flood test is not enabled Merge branch 'topic/hda-bus-ops-cleanup' of https://git.kernel.org/.../tiwai/sound into asoc-5.4 Merge branch 'topic/hda-bus-ops-cleanup' of https://git.kernel.org/.../tiwai/sound into asoc-5.4 Merge branch 'asoc-5.3' into asoc-5.4 Merge branch 'asoc-5.3' into asoc-linus Merge branch 'asoc-5.4' into asoc-next Masanari Iida (1): ASoC: ti: Fix typos in ti/Kconfig Mauro Carvalho Chehab (1): ASoC: sunxi: fix a sound binding broken reference Maxime Ripard (28): ASoC: sun4i-i2s: Register regmap and PCM before our component ASoC: sun4i-i2s: Switch to devm for PCM register ASoC: sun4i-i2s: Don't use the oversample to calculate BCLK ASoC: sun4i-i2s: Replace call to params_channels by local variable ASoC: sun4i-i2s: Rework MCLK divider calculation ASoC: sun4i-i2s: Move the channel configuration to a callback ASoC: sun4i-i2s: Move the format configuration to a callback ASoC: sun4i-i2s: Use module clock as BCLK parent on newer SoCs ASoC: sun4i-i2s: Fix the MCLK and BCLK dividers on newer SoCs ASoC: sun4i-i2s: RX and TX counter registers are swapped ASoC: sun4i-i2s: Fix LRCK and BCLK polarity offsets on newer SoCs ASoC: sun4i-i2s: Fix the LRCK polarity ASoC: sun4i-i2s: Fix WSS and SR fields for the A83t ASoC: sun4i-i2s: Fix MCLK Enable bit offset on A83t ASoC: sun4i-i2s: Fix the LRCK period on A83t ASoC: sun4i-i2s: Remove duplicated quirks structure ASoC: sun4i-i2s: Pass the channels number as an argument ASoC: sun4i-i2s: Support more channels ASoC: sun4i-i2s: Add support for TDM slots ASoC: sun4i-i2s: Use the physical / slot width for the clocks ASoC: sun4i-i2s: Use the actual format width instead of an hardcoded one ASoC: sun4i-i2s: Replace call to params_width by local variable ASoC: sun4i-i2s: Add support for DSP formats ASoC: sunxi: Revert initial A83t support ASoC: sun4i: Revert A83t description ASoC: dt-bindings: sun4i-spdif: Fix dma-names warning ASoC: dt-bindings: Convert Allwinner A33 codec to a schema ASoC: dt-bindings: Convert Allwinner A64 analog codec to a schema Michał Mirosław (5): ASoC: wm8904: fix typo in DAPM kcontrol name ASoC: wm8904: implement input mode select as a mux ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig ASoC: atmel_ssc_dai: rework DAI format configuration ASoC: atmel_ssc_dai: implement left-justified data mode Oleksandr Suvorov (6): ASoC: Define a set of DAPM pre/post-up events ASoC: sgtl5000: Improve VAG power and mute control ASoC: sgtl5000: Fix definition of VAG Ramp Control ASoC: sgtl5000: add ADC mute control ASoC: sgtl5000: Fix of unmute outputs on probe ASoC: sgtl5000: Fix charge pump source assignment Pan Xiuli (5): ASoC: SOF: pci: mark last_busy value at runtime PM init ASoC: Intel: common: add ACPI matching tables for Tiger Lake ASoC: SOF: Intel: initial support for Tiger Lake. ASoC: Intel: common: add ACPI matching tables for EHL ASoC: SOF: Intel: initial support for Elkhart Lake Peter Ujfalusi (12): ASoC: pcm3168a: Allow all channels in case of parallel DIN/DOUT setup ASoC: ti: davinci-mcasp: Add support for RIGHT_J format ASoC: ti: davinci-mcasp: Improve serializer handling in multi AXR setups ASoC: ti: davinci-mcasp: Support for correct symmetric sample bits ASoC: pcm3168a: Retain the independence of DAC and ADC side of the codec ASoC: pcm3168a: Allow reconfiguration of tdm_slots and slot_width ASoC: core: Move pcm_mutex up to card level from snd_soc_pcm_runtime ASoC: ti: davinci-i2s: Move the XSYNCERR workaround to .prepare callback ASoC: ti: davinci-i2s: Add S32_LE as support format ASoC: ti: edma-pcm: Fix for legacy dma_slave_map based channel lookup ASoC: ti: davinci-evm: Do not fail if the dai_set_sysclk returns -ENOTSUPP ASoC: dmaengine: Make the pcm->name equal to pcm->id if the name is not set Pierre-Louis Bossart (14): ASoC: Intel: Skylake: move NHLT header to common directory ALSA: hda: move parts of NHLT code to new module ALSA: hda: intel-nhlt: handle NHLT VENDOR_DEFINED DMIC geometry ASoC: Intel: Skylake: use common NHLT module ALSA: hda/intel: stop probe if DMICS are detected on Skylake+ platforms ASoC: SOF: Intel: hda: fix MSI handling ASoC: soc-core: fix module_put() warning in soc_cleanup_component ASoC: SOF: Intel: hda: fixup HDaudio topology name with DMIC number ASoC: Intel: hsw: remove i386 build warning w/ size_t argument ASoC: SOF: fix HDA direct MMIO access ASoC: SOF: ipc: add ALH parameters ASoC: SOF: topology: initial support for Intel ALH DAI type ASoC: SOF: pcm: add ALH support ASoC: SOF/Intel: fix selection of SND_INTEL_NHTL Rander Wang (4): ASoC: SOF: Intel: hda: reduce ifdef usage for hda ASoC: SOF: Intel: hda: Enable jack detection ASoC: SOF: Intel: hda: fix link DMA config ASoC: SOF: Intel: hda: fix stream id setting Ranjani Sridharan (2): ASoC: SOF: Intel: hda: Make hdac_device device-managed ASoC: SOF: Intel: hda: Initialize HDA controller after i915 init Richard Fitzgerald (1): ASoC: cs47l15: Add codec driver for Cirrus Logic CS47L15 Sathya Prakash M R (1): ASoC: Intel: boards: Match Product Family instead of product Shengjiu Wang (8): ASoC: fsl_esai: Wrap some operations to be functions ASoC: fsl_esai: recover the channel swap after xrun ASoC: fsl_esai: Add compatible string for imx6ull ASoC: fsl_esai: Add new compatible string for imx6ull ASoC: imx-audmux: Add driver suspend and resume to support MEGA Fast ASoC: imx-audmix: register the card on a proper dev ASoC: cs42xx8: Force suspend/resume during system suspend/resume ASoC: fsl_ssi: Fix clock control issue in master mode Shuming Fan (6): ASoC: rt1308: simplify the EFUSE read function ASoC: rt1308: add silence detection and manual PDB control ASoC: rt1308: Convert headers to SPDX ASoC: rt1011: Add R0 temperature and TDM1 ADC2DAT Swap control ASoC: rt1011: Add a flag for the R0 calibration test ASoC: rt1011: add mutex protection to set_fmt/set_tdm_slot Srinivas Kandagatla (1): ASoC: wcd9335: Fix primary interpolator max rate Stephen Boyd (1): ASoC: Remove dev_err() usage after platform_get_irq() Stephen Rothwell (1): ALSA: hda: readl/writel need linux/io.h Stuart Henderson (1): ASoC: cs47l92: Add codec driver for Cirrus Logic CS47L92 Szymon Mielczarek (1): ASoC: dapm: Invalidate only paths reachable for a given stream Takashi Iwai (4): ALSA: hda: Remove page allocation redirection ALSA: hda: Direct MMIO accesses ASoC: SOF: Intel: Initialize hdaudio bus properly ASoC: hdac_hdmi: Offload dapm update at jack detection Timo Wischer (1): ASoC: rsnd: Support hw_free() callback at DAI level Tzung-Bi Shih (6): ASoC: mediatek: mt8183: make headset codec optional ASoC: mediatek: mt8183: remove unused DAPM-related variables ASoC: mediatek: mt8183: move private structure ASoC: mediatek: mt8183: move headset jack to card-specific storage ASoC: mediatek: mt8183: remove forward declaration of headset_init ASoC: mediatek: mt8183: fix space issues Vidyakumar Athota (1): ALSA: pcm: add support for 352.8KHz and 384KHz sample rate Wei Yongjun (1): ASoC: SOF: imx8: Fix return value check in imx8_probe() YueHaibing (66): ASoC: tegra20_das: use devm_platform_ioremap_resource() to simplify code ASoC: tegra: use devm_platform_ioremap_resource() to simplify code ASoC: rockchip: use devm_platform_ioremap_resource() to simplify code ASoC: ep93xx-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: mt8173: use devm_platform_ioremap_resource() to simplify code ASoC: mt6797: use devm_platform_ioremap_resource() to simplify code ASoC: imx-audmux: use devm_platform_ioremap_resource() to simplify code ASoC: fsl_audmix: use devm_platform_ioremap_resource() to simplify code ASoC: bcm2835-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: sirf: use devm_platform_ioremap_resource() to simplify code ASoC: mxs-saif: use devm_platform_ioremap_resource() to simplify code ASoC: spear: use devm_platform_ioremap_resource() to simplify code ASoC: kirkwood-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: xtfpga-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-pdm: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-spdifin: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-spdifout: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-fifo: use devm_platform_ioremap_resource() to simplify code ASoC: xlnx: use devm_platform_ioremap_resource() to simplify code ASoC: sun8i-codec-analog: use devm_platform_ioremap_resource() to simplify code ASoC: sunxi: sun50i-codec-analog: use devm_platform_ioremap_resource() to simplify code ASoC: sun8i-codec: use devm_platform_ioremap_resource() to simplify code ASoC: stm32: sai: use devm_platform_ioremap_resource() to simplify code ASoC: codecs: msm8916-wcd: use devm_platform_ioremap_resource() to simplify code ASoC: codecs: jz4725b: use devm_platform_ioremap_resource() to simplify code ASoC: mmp-sspa: use devm_platform_ioremap_resource() to simplify code ASoC: jz4740: use devm_platform_ioremap_resource() to simplify code ASoC: inno_rk3036: use devm_platform_ioremap_resource() to simplify code ASoC: uniphier: evea: use devm_platform_ioremap_resource() to simplify code ASoC: uniphier: aio-dma: use devm_platform_ioremap_resource() to simplify code ASoC: psc-ac97: use devm_platform_ioremap_resource() to simplify code ASoC: au1x: psc-i2s: use devm_platform_ioremap_resource() to simplify code ASoC: meson: g12a-tohdmitx: use devm_platform_ioremap_resource() to simplify code ASoC: meson: axg-tdm-formatter: use devm_platform_ioremap_resource() to simplify code ASoC: tscs454: remove unused variable 'PLL_48K_RATE' ASoC: SOF: Intel: Add missing include file hdac_hda.h ASoC: mt6351: remove unused variable 'mt_lineout_control' ASoC: ml26124: remove unused variable 'ngth' ASoC: max98926: remove two unused variables ASoC: max9850: remove unused variable 'max9850_reg' ASoC: es8328: Fix copy-paste error in es8328_right_line_controls ASoC: cs4349: Use PM ops 'cs4349_runtime_pm' ASoC: wm8737: Fix copy-paste error in wm8737_snd_controls ASoC: soc-core: Fix -Wunused-const-variable warning ASoC: mt2701: remove unused variables ASoC: mediatek: mt8183-da7219-max98357: remove unused variable ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: remove unused variables ASoC: cs42l73: remove unused variables 'vsp_output_mux' and 'xsp_output_mux' ASoC: cs4349: Use PM ops 'cs4349_runtime_pm' ASoC: max98371: remove unused variable 'max98371_noload_gain_tlv' ASoC: rt1011: remove unused variable 'dac_vol_tlv' and 'adc_vol_tlv' ASoC: tlv320aic23: remove unused variable 'tlv320aic23_rec_src' ASoC: tlv320aic31xx: remove unused variable 'cm_m_enum' ASoC: es8328: remove unused variable 'pga_tlv' ASoC: es8328: Fix copy-paste error in es8328_right_line_controls ASoC: cs42l56: remove unused variable 'adc_swap_enum' ASoC: 88pm860x: remove unused variables 'pcm_switch_controls' and 'aif1_mux' ASoC: AMD: Fix Kconfig warning without GPIOLIB ASoC: wm8988: fix typo in wm8988_right_line_controls ASoC: mediatek: mt2701: Fix -Wunused-const-variable warnings ASoC: SOF: imx8: Make some functions static ASoC: amd: use devm_platform_ioremap_resource() to simplify code ASoC: ep93xx: use devm_platform_ioremap_resource() to simplify code ASoC: sirf-audio: use devm_platform_ioremap_resource() to simplify code ASoC: tegra: use devm_platform_ioremap_resource() to simplify code ASoC: SOF: imx8: Fix COMPILE_TEST error Zhu Yingjiang (3): ASoC: SOF: Intel: hda: set position buffer in init chip ASoC: SOF: Intel: hda: use SOF defined init chip in resume ASoC: SOF: Intel: hda: remove duplicated clear WAKESTS fengchunguo (1): ASoC: max98383: fix i2c probe failure Documentation/devicetree/bindings/dsp/fsl,dsp.yaml | 88 + .../bindings/sound/allwinner,sun4i-a10-spdif.yaml | 4 +- .../sound/allwinner,sun50i-a64-codec-analog.yaml | 39 + .../bindings/sound/allwinner,sun8i-a33-codec.yaml | 57 + .../devicetree/bindings/sound/amlogic,axg-fifo.txt | 9 +- .../devicetree/bindings/sound/amlogic,axg-pdm.txt | 6 +- .../bindings/sound/amlogic,axg-spdifin.txt | 6 +- .../bindings/sound/amlogic,axg-spdifout.txt | 6 +- .../bindings/sound/amlogic,axg-tdm-formatters.txt | 4 +- .../bindings/sound/amlogic,g12a-tohdmitx.txt | 5 +- .../devicetree/bindings/sound/everest,es8316.txt | 23 + .../devicetree/bindings/sound/fsl,esai.txt | 7 +- .../devicetree/bindings/sound/fsl-sai.txt | 4 +- .../bindings/sound/sun50i-codec-analog.txt | 14 - .../devicetree/bindings/sound/sun8i-a33-codec.txt | 63 - .../devicetree/bindings/sound/uda1334.txt | 17 + arch/arm/mach-omap1/board-ams-delta.c | 10 +- include/sound/hdaudio.h | 69 +- include/sound/hdaudio_ext.h | 1 - include/sound/hdmi-codec.h | 17 + .../skl-nhlt.h => include/sound/intel-nhlt.h | 51 +- include/sound/pcm.h | 5 + include/sound/soc-acpi-intel-match.h | 2 + include/sound/soc-component.h | 387 ++++ include/sound/soc-dai.h | 28 +- include/sound/soc-dapm.h | 14 +- include/sound/soc-dpcm.h | 9 +- include/sound/soc.h | 348 +--- include/sound/sof/dai-intel.h | 12 + include/sound/sof/dai.h | 5 +- include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 9 + sound/core/pcm_native.c | 2 +- sound/hda/Kconfig | 8 + sound/hda/Makefile | 3 + sound/hda/ext/hdac_ext_bus.c | 60 +- sound/hda/hdac_bus.c | 37 +- sound/hda/hdac_controller.c | 18 +- sound/hda/hdac_stream.c | 8 +- sound/hda/intel-nhlt.c | 107 + sound/pci/hda/Kconfig | 11 + sound/pci/hda/hda_codec.c | 8 +- sound/pci/hda/hda_controller.c | 6 +- sound/pci/hda/hda_controller.h | 3 +- sound/pci/hda/hda_intel.c | 105 +- sound/pci/hda/hda_tegra.c | 84 +- sound/soc/Kconfig | 1 - sound/soc/Makefile | 3 +- sound/soc/amd/Kconfig | 2 +- sound/soc/amd/acp-pcm-dma.c | 3 +- sound/soc/atmel/Kconfig | 30 +- sound/soc/atmel/atmel-classd.c | 7 +- sound/soc/atmel/atmel-pdmic.c | 7 +- sound/soc/atmel/atmel_ssc_dai.c | 293 +-- sound/soc/atmel/mchp-i2s-mcc.c | 111 +- sound/soc/au1x/psc-ac97.c | 5 +- sound/soc/au1x/psc-i2s.c | 5 +- sound/soc/bcm/bcm2835-i2s.c | 4 +- sound/soc/bcm/cygnus-pcm.c | 6 +- sound/soc/bcm/cygnus-ssp.c | 7 +- sound/soc/cirrus/ep93xx-ac97.c | 4 +- sound/soc/cirrus/ep93xx-i2s.c | 4 +- sound/soc/codecs/88pm860x-codec.c | 15 - sound/soc/codecs/Kconfig | 21 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ad193x.c | 19 +- sound/soc/codecs/cs4271.c | 6 +- sound/soc/codecs/cs42l56.c | 8 - sound/soc/codecs/cs42l73.c | 6 - sound/soc/codecs/cs42xx8.c | 2 + sound/soc/codecs/cs4349.c | 1 + sound/soc/codecs/cs47l15.c | 1490 ++++++++++++++ sound/soc/codecs/cs47l35.c | 2 +- sound/soc/codecs/cs47l90.c | 9 +- sound/soc/codecs/cs47l92.c | 2039 ++++++++++++++++++++ sound/soc/codecs/es8316.c | 73 +- sound/soc/codecs/es8328.c | 3 +- sound/soc/codecs/hdac_hda.c | 4 + sound/soc/codecs/hdac_hdmi.c | 57 +- sound/soc/codecs/hdmi-codec.c | 46 + sound/soc/codecs/inno_rk3036.c | 4 +- sound/soc/codecs/jz4725b.c | 4 +- sound/soc/codecs/jz4740.c | 4 +- sound/soc/codecs/madera.c | 531 ++++- sound/soc/codecs/madera.h | 10 + sound/soc/codecs/max98371.c | 4 - sound/soc/codecs/max98373.c | 34 +- sound/soc/codecs/max98373.h | 1 + sound/soc/codecs/max9850.c | 13 - sound/soc/codecs/max98926.c | 9 - sound/soc/codecs/ml26124.c | 1 - sound/soc/codecs/msm8916-wcd-analog.c | 12 +- sound/soc/codecs/msm8916-wcd-digital.c | 4 +- sound/soc/codecs/mt6351.c | 5 - sound/soc/codecs/mt6358.c | 10 +- sound/soc/codecs/pcm3168a.c | 133 +- sound/soc/codecs/rk3328_codec.c | 4 +- sound/soc/codecs/rt1011.c | 56 +- sound/soc/codecs/rt1011.h | 3 +- sound/soc/codecs/rt1305.c | 3 +- sound/soc/codecs/rt1308.c | 51 +- sound/soc/codecs/rt1308.h | 6 +- sound/soc/codecs/rt5665.c | 8 +- sound/soc/codecs/rt5677.c | 20 +- sound/soc/codecs/sgtl5000.c | 248 ++- sound/soc/codecs/sgtl5000.h | 2 +- sound/soc/codecs/sirf-audio-codec.c | 4 +- sound/soc/codecs/tlv320aic23.c | 2 - sound/soc/codecs/tlv320aic31xx.c | 8 +- sound/soc/codecs/tscs454.c | 1 - sound/soc/codecs/twl6040.c | 4 +- sound/soc/codecs/uda1334.c | 295 +++ sound/soc/codecs/wcd-clsh-v2.c | 2 +- sound/soc/codecs/wcd9335.c | 23 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8904.c | 56 +- sound/soc/codecs/wm8955.c | 4 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm_adsp.c | 6 +- sound/soc/codecs/wm_adsp.h | 4 +- sound/soc/fsl/fsl_asrc.c | 4 +- sound/soc/fsl/fsl_audmix.c | 4 +- sound/soc/fsl/fsl_esai.c | 267 ++- sound/soc/fsl/fsl_sai.c | 358 ++-- sound/soc/fsl/fsl_sai.h | 85 +- sound/soc/fsl/fsl_spdif.c | 4 +- sound/soc/fsl/fsl_ssi.c | 22 +- sound/soc/fsl/imx-audmix.c | 4 +- sound/soc/fsl/imx-audmux.c | 58 +- sound/soc/fsl/imx-ssi.c | 4 +- sound/soc/generic/audio-graph-card.c | 19 + sound/soc/generic/simple-card-utils.c | 7 + sound/soc/generic/simple-card.c | 22 +- sound/soc/intel/Kconfig | 1 + sound/soc/intel/baytrail/sst-baytrail-ipc.c | 65 +- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 1 + sound/soc/intel/boards/Kconfig | 28 +- sound/soc/intel/boards/bdw-rt5677.c | 6 +- sound/soc/intel/boards/broadwell.c | 6 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 31 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 101 +- sound/soc/intel/boards/haswell.c | 6 +- sound/soc/intel/boards/skl_hda_dsp_common.c | 5 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 + sound/soc/intel/boards/sof_rt5682.c | 7 +- sound/soc/intel/common/Makefile | 1 + sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 12 + sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 18 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 24 + sound/soc/intel/common/soc-intel-quirks.h | 5 + sound/soc/intel/common/sst-acpi.c | 3 +- sound/soc/intel/common/sst-ipc.c | 71 +- sound/soc/intel/common/sst-ipc.h | 28 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 164 +- sound/soc/intel/skylake/Makefile | 12 +- sound/soc/intel/skylake/bxt-sst.c | 50 +- sound/soc/intel/skylake/cnl-sst-dsp.h | 7 +- sound/soc/intel/skylake/cnl-sst.c | 43 +- sound/soc/intel/skylake/skl-debug.c | 44 +- sound/soc/intel/skylake/skl-messages.c | 264 ++- sound/soc/intel/skylake/skl-nhlt.c | 109 +- sound/soc/intel/skylake/skl-pcm.c | 74 +- sound/soc/intel/skylake/skl-ssp-clk.c | 5 +- sound/soc/intel/skylake/skl-sst-dsp.c | 10 +- sound/soc/intel/skylake/skl-sst-dsp.h | 29 +- sound/soc/intel/skylake/skl-sst-ipc.c | 160 +- sound/soc/intel/skylake/skl-sst-ipc.h | 55 +- sound/soc/intel/skylake/skl-sst-utils.c | 37 +- sound/soc/intel/skylake/skl-sst.c | 54 +- sound/soc/intel/skylake/skl-topology.c | 494 ++--- sound/soc/intel/skylake/skl-topology.h | 43 +- sound/soc/intel/skylake/skl.c | 73 +- sound/soc/intel/skylake/skl.h | 105 +- sound/soc/kirkwood/kirkwood-i2s.c | 8 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +- sound/soc/mediatek/common/mtk-btcvsd.c | 4 +- sound/soc/mediatek/mt2701/mt2701-afe-common.h | 21 - sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 38 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 5 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 8 +- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 15 +- sound/soc/mediatek/mt8183/mt8183-dai-tdm.c | 177 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 150 +- sound/soc/mediatek/mt8183/mt8183-reg.h | 8 +- sound/soc/meson/Kconfig | 1 + sound/soc/meson/axg-card.c | 11 +- sound/soc/meson/axg-fifo.c | 6 +- sound/soc/meson/axg-fifo.h | 1 + sound/soc/meson/axg-frddr.c | 105 +- sound/soc/meson/axg-pdm.c | 4 +- sound/soc/meson/axg-spdifin.c | 4 +- sound/soc/meson/axg-spdifout.c | 4 +- sound/soc/meson/axg-tdm-formatter.c | 6 +- sound/soc/meson/axg-tdmin.c | 47 +- sound/soc/meson/axg-tdmout.c | 103 +- sound/soc/meson/axg-toddr.c | 83 +- sound/soc/meson/g12a-tohdmitx.c | 38 +- sound/soc/mxs/mxs-saif.c | 13 +- sound/soc/nuc900/Kconfig | 29 - sound/soc/nuc900/Makefile | 12 - sound/soc/nuc900/nuc900-ac97.c | 391 ---- sound/soc/nuc900/nuc900-audio.c | 73 - sound/soc/nuc900/nuc900-audio.h | 108 -- sound/soc/nuc900/nuc900-pcm.c | 321 --- sound/soc/pxa/mmp-sspa.c | 4 +- sound/soc/qcom/common.c | 22 +- sound/soc/qcom/lpass-platform.c | 5 +- sound/soc/rockchip/rockchip_i2s.c | 3 + sound/soc/rockchip/rockchip_max98090.c | 18 +- sound/soc/samsung/neo1973_wm8753.c | 3 +- sound/soc/samsung/speyside.c | 3 +- sound/soc/samsung/tm2_wm5110.c | 10 +- sound/soc/sh/rcar/adg.c | 21 +- sound/soc/sh/rcar/core.c | 16 +- sound/soc/sh/rcar/rsnd.h | 12 +- sound/soc/sirf/sirf-usp.c | 4 +- sound/soc/soc-component.c | 561 ++++++ sound/soc/soc-compress.c | 57 +- sound/soc/soc-core.c | 1269 +++++------- sound/soc/soc-dai.c | 407 ++++ sound/soc/soc-dapm.c | 361 ++-- sound/soc/soc-generic-dmaengine-pcm.c | 6 + sound/soc/soc-jack.c | 18 - sound/soc/soc-pcm.c | 546 ++---- sound/soc/soc-topology.c | 8 +- sound/soc/soc-utils.c | 199 -- sound/soc/sof/Kconfig | 11 + sound/soc/sof/Makefile | 8 +- sound/soc/sof/core.c | 4 +- sound/soc/sof/debug.c | 50 +- sound/soc/sof/imx/Kconfig | 23 + sound/soc/sof/imx/Makefile | 4 + sound/soc/sof/imx/imx8.c | 394 ++++ sound/soc/sof/intel/Kconfig | 33 + sound/soc/sof/intel/apl.c | 4 +- sound/soc/sof/intel/bdw.c | 166 +- sound/soc/sof/intel/byt.c | 174 +- sound/soc/sof/intel/cnl.c | 36 +- sound/soc/sof/intel/hda-bus.c | 86 +- sound/soc/sof/intel/hda-codec.c | 50 +- sound/soc/sof/intel/hda-ctrl.c | 21 +- sound/soc/sof/intel/hda-dai.c | 38 +- sound/soc/sof/intel/hda-dsp.c | 100 +- sound/soc/sof/intel/hda-ipc.c | 150 +- sound/soc/sof/intel/hda.c | 132 +- sound/soc/sof/intel/hda.h | 18 +- sound/soc/sof/ipc.c | 8 +- sound/soc/sof/loader.c | 182 +- sound/soc/sof/ops.h | 46 +- sound/soc/sof/pcm.c | 66 +- sound/soc/sof/pm.c | 6 +- sound/soc/sof/sof-of-dev.c | 143 ++ sound/soc/sof/sof-pci-dev.c | 47 + sound/soc/sof/sof-priv.h | 19 +- sound/soc/sof/topology.c | 275 ++- sound/soc/sof/trace.c | 9 +- sound/soc/spear/spdif_in.c | 5 +- sound/soc/sprd/sprd-mcdt.c | 4 +- sound/soc/sti/sti_uniperif.c | 4 +- sound/soc/stm/stm32_i2s.c | 5 +- sound/soc/stm/stm32_sai.c | 8 +- sound/soc/stm/stm32_spdifrx.c | 4 +- sound/soc/sunxi/sun4i-codec.c | 14 +- sound/soc/sunxi/sun4i-i2s.c | 671 ++++--- sound/soc/sunxi/sun50i-codec-analog.c | 4 +- sound/soc/sunxi/sun8i-codec-analog.c | 4 +- sound/soc/sunxi/sun8i-codec.c | 4 +- sound/soc/tegra/tegra20_das.c | 4 +- sound/soc/tegra/tegra30_ahub.c | 5 +- sound/soc/tegra/tegra30_i2s.c | 4 +- sound/soc/ti/Kconfig | 4 +- sound/soc/ti/ams-delta.c | 31 +- sound/soc/ti/davinci-evm.c | 2 +- sound/soc/ti/davinci-i2s.c | 90 +- sound/soc/ti/davinci-mcasp.c | 185 +- sound/soc/ti/edma-pcm.c | 17 +- sound/soc/ti/n810.c | 1 + sound/soc/ti/rx51.c | 15 +- sound/soc/uniphier/aio-cpu.c | 31 +- sound/soc/uniphier/aio-dma.c | 8 +- sound/soc/uniphier/aio.h | 1 + sound/soc/uniphier/evea.c | 4 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 - sound/soc/xilinx/xlnx_i2s.c | 4 +- sound/soc/xilinx/xlnx_spdif.c | 3 +- sound/soc/xtensa/xtfpga-i2s.c | 5 +- sound/soc/zte/zx-tdm.c | 1 - 287 files changed, 12569 insertions(+), 7014 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/fsl,dsp.yaml create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-a64-codec-analog.yaml create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun8i-a33-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/everest,es8316.txt delete mode 100644 Documentation/devicetree/bindings/sound/sun50i-codec-analog.txt delete mode 100644 Documentation/devicetree/bindings/sound/sun8i-a33-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/uda1334.txt rename sound/soc/intel/skylake/skl-nhlt.h => include/sound/intel-nhlt.h (65%) create mode 100644 include/sound/soc-component.h create mode 100644 sound/hda/intel-nhlt.c create mode 100644 sound/soc/codecs/cs47l15.c create mode 100644 sound/soc/codecs/cs47l92.c create mode 100644 sound/soc/codecs/uda1334.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-ehl-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-tgl-match.c delete mode 100644 sound/soc/nuc900/Kconfig delete mode 100644 sound/soc/nuc900/Makefile delete mode 100644 sound/soc/nuc900/nuc900-ac97.c delete mode 100644 sound/soc/nuc900/nuc900-audio.c delete mode 100644 sound/soc/nuc900/nuc900-audio.h delete mode 100644 sound/soc/nuc900/nuc900-pcm.c create mode 100644 sound/soc/soc-component.c create mode 100644 sound/soc/soc-dai.c create mode 100644 sound/soc/sof/imx/Kconfig create mode 100644 sound/soc/sof/imx/Makefile create mode 100644 sound/soc/sof/imx/imx8.c create mode 100644 sound/soc/sof/sof-of-dev.c

2 1

[alsa-devel] [PATCH] ASoC: SOF: Intel: work around snd_hdac_aligned_read link failure
by Arnd Bergmann 10 Sep '19

10 Sep '19

When CONFIG_SND_HDA_ALIGNED_MMIO is selected by another driver (i.e. Tegra) that selects CONFIG_SND_HDA_CORE as a loadable module, but SND_SOC_SOF_HDA_COMMON is built-in, we get a link failure from some functions that access the hda register: sound/soc/sof/intel/hda.o: In function `hda_ipc_irq_dump': hda.c:(.text+0x784): undefined reference to `snd_hdac_aligned_read' sound/soc/sof/intel/hda-stream.o: In function `hda_dsp_stream_threaded_handler': hda-stream.c:(.text+0x12e4): undefined reference to `snd_hdac_aligned_read' hda-stream.c:(.text+0x12f8): undefined reference to `snd_hdac_aligned_write' Add an explicit 'select' statement as a workaround. This is not a great solution, but it's the easiest way I could come up with. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sof/intel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..9180184026e1 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -248,6 +248,7 @@ config SND_SOC_SOF_HDA_COMMON tristate select SND_SOC_SOF_INTEL_COMMON select SND_SOC_SOF_HDA_LINK_BASELINE + select SND_HDA_CORE if SND_HDA_ALIGNED_MMIO help This option is not user-selectable but automagically handled by 'select' statements at a higher level -- 2.20.0

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[alsa-devel] omap-mcbsp: TX Buffer Overflow
by Tomas Novotny 10 Sep '19

10 Sep '19

Hi, we have AM3703 based board similar to BeagleBoard. I'm hitting this error after upgrade to latest LTS 4.19.71 (upgraded from 4.1): omap-mcbsp 49022000.mcbsp: TX Buffer Overflow! This appears during or after playing of short (~2s) ding-dong wav. That error exists for longer time, because handling of tx buffer overflow irq was introduced in 2016: 4e85e7776eba ("ASoC: omap-mcbsp: Enable TX/RX under and overflow interrupts"). I've cherry-picked it to 4.1 and I see the error there also. The sound seems clear and ok to me, but we are using low quality speaker. There are two workarounds to get rid of the message: 1) Change 'dma_op_mode' sysfs attribute from 'element' to 'threshold'. I found that just by coincidence when checking sysfs attributes. 2) Compile kernel with CONFIG_VIDEO_OMAP3=y. Found on Logic PD forum [1]. Does anybody have any idea what's going wrong? Or why these (somehow) unrelated workarounds help? Thanks, Tomas [1] https://support.logicpd.com/TDGForum/tabid/124/aft/2277/Default.aspx

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[alsa-devel] [PATCH] ASoC: fsl_sai: Fix noise when using EDMA
by Daniel Baluta 09 Sep '19

09 Sep '19

From: Mihai Serban <mihai.serban(a)nxp.com> EDMA requires the period size to be multiple of maxburst. Otherwise the remaining bytes are not transferred and thus noise is produced. We can handle this issue by adding a constraint on SNDRV_PCM_HW_PARAM_PERIOD_SIZE to be multiple of tx/rx maxburst value. Cc: NXP Linux Team <linux-imx(a)nxp.com> Signed-off-by: Mihai Serban <mihai.serban(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 15 +++++++++++++++ sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 16 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 728307acab90..fe126029f4e3 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -612,6 +612,16 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE); + /* + * some DMA controllers need period size to be a multiple of + * tx/rx maxburst + */ + if (sai->soc_data->use_constraint_period_size) + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + tx ? sai->dma_params_tx.maxburst : + sai->dma_params_rx.maxburst); + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &fsl_sai_rate_constraints); @@ -1011,30 +1021,35 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .use_imx_pcm = false, .fifo_depth = 32, .reg_offset = 0, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .use_imx_pcm = true, .fifo_depth = 32, .reg_offset = 0, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .use_imx_pcm = true, .fifo_depth = 16, .reg_offset = 8, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .use_imx_pcm = true, .fifo_depth = 128, .reg_offset = 8, + .use_constraint_period_size = false, }; static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .use_imx_pcm = true, .fifo_depth = 64, .reg_offset = 0, + .use_constraint_period_size = true, }; static const struct of_device_id fsl_sai_ids[] = { diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index b89b0ca26053..3a3f6f8e5595 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -157,6 +157,7 @@ struct fsl_sai_soc_data { bool use_imx_pcm; + bool use_constraint_period_size; unsigned int fifo_depth; unsigned int reg_offset; }; -- 2.17.1

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[alsa-devel] about pop chime on pcm api
by Yang, Xiaotao (X.) 09 Sep '19

09 Sep '19

Hi: Why should I call api<snd_pcm_close()> between the interval if I want to play a wav file two times repeatly, Otherwise pop chime occurs? Eg: (fake code as follows): Following codes works good. Pcm.open();//open pcm device Pcm.play(demo.wav);//set parameter and call snd_pcm_writei() to play Pcm.close();//close pcm device Pcm.open(); Pcm.play(demo.wav); Pcm.close(); But following codes works with pop chime Pcm.open(); Pcm.play(demo.wav); Pcm.play(demo.wav); Pcm.close();

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[alsa-devel] Applied "ASoC: hdmi-codec: Add an op to set callback function for plug event" to the asoc tree
by Mark Brown 09 Sep '19

09 Sep '19

The patch ASoC: hdmi-codec: Add an op to set callback function for plug event has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6fa5963c37a2e3335eba0b7455e35a01318ebc15 Mon Sep 17 00:00:00 2001 From: Cheng-Yi Chiang <cychiang(a)chromium.org> Date: Wed, 17 Jul 2019 16:33:23 +0800 Subject: [PATCH] ASoC: hdmi-codec: Add an op to set callback function for plug event Add an op in hdmi_codec_ops so codec driver can register callback function to handle plug event. Driver in DRM can use this callback function to report connector status. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Link: https://lore.kernel.org/r/20190717083327.47646-2-cychiang@chromium.org Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/hdmi-codec.h | 17 +++++++++++++ sound/soc/codecs/hdmi-codec.c | 46 +++++++++++++++++++++++++++++++++++ 2 files changed, 63 insertions(+) diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 7fea496f1f34..83b17682e01c 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -47,6 +47,9 @@ struct hdmi_codec_params { int channels; }; +typedef void (*hdmi_codec_plugged_cb)(struct device *dev, + bool plugged); + struct hdmi_codec_pdata; struct hdmi_codec_ops { /* @@ -88,6 +91,14 @@ struct hdmi_codec_ops { */ int (*get_dai_id)(struct snd_soc_component *comment, struct device_node *endpoint); + + /* + * Hook callback function to handle connector plug event. + * Optional + */ + int (*hook_plugged_cb)(struct device *dev, void *data, + hdmi_codec_plugged_cb fn, + struct device *codec_dev); }; /* HDMI codec initalization data */ @@ -99,6 +110,12 @@ struct hdmi_codec_pdata { void *data; }; +struct snd_soc_component; +struct snd_soc_jack; + +int hdmi_codec_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + #define HDMI_CODEC_DRV_NAME "hdmi-audio-codec" #endif /* __HDMI_CODEC_H__ */ diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 0bf1c8cad108..b5fd8f08726e 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -7,6 +7,7 @@ #include <linux/module.h> #include <linux/string.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -274,6 +275,8 @@ struct hdmi_codec_priv { struct snd_pcm_chmap *chmap_info; unsigned int chmap_idx; struct mutex lock; + struct snd_soc_jack *jack; + unsigned int jack_status; }; static const struct snd_soc_dapm_widget hdmi_widgets[] = { @@ -663,6 +666,49 @@ static int hdmi_dai_probe(struct snd_soc_dai *dai) return 0; } +static void hdmi_codec_jack_report(struct hdmi_codec_priv *hcp, + unsigned int jack_status) +{ + if (hcp->jack && jack_status != hcp->jack_status) { + snd_soc_jack_report(hcp->jack, jack_status, SND_JACK_LINEOUT); + hcp->jack_status = jack_status; + } +} + +static void plugged_cb(struct device *dev, bool plugged) +{ + struct hdmi_codec_priv *hcp = dev_get_drvdata(dev); + + if (plugged) + hdmi_codec_jack_report(hcp, SND_JACK_LINEOUT); + else + hdmi_codec_jack_report(hcp, 0); +} + +/** + * hdmi_codec_set_jack_detect - register HDMI plugged callback + * @component: the hdmi-codec instance + * @jack: ASoC jack to report (dis)connection events on + */ +int hdmi_codec_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); + int ret = -EOPNOTSUPP; + + if (hcp->hcd.ops->hook_plugged_cb) { + hcp->jack = jack; + ret = hcp->hcd.ops->hook_plugged_cb(component->dev->parent, + hcp->hcd.data, + plugged_cb, + component->dev); + if (ret) + hcp->jack = NULL; + } + return ret; +} +EXPORT_SYMBOL_GPL(hdmi_codec_set_jack_detect); + static int hdmi_dai_spdif_probe(struct snd_soc_dai *dai) { struct hdmi_codec_daifmt *cf = dai->playback_dma_data; -- 2.20.1

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[alsa-devel] Alsa conformance test - a new test for ALSA drivers
by Yu-Hsuan Hsu 09 Sep '19

09 Sep '19

Hi all, I have created an new tool "Alsa conformance test"( https://chromium.googlesource.com/chromiumos/platform/audiotest/+/master/al…), which is used for verifying ALSA drivers. Here are some main functions it can measure: 1. Correctness of hw_params 2. Stability of rate 3. Stability of step 4. Runtime of each ALSA API 5. Underrun and Overrun I recommend to use the script "alsa_conformance_test.py" inside. Just give it a device and then it will test the below functions automatically. 1. To make sure whether all the parameters provided by an ALSA driver can be used. 2. To make sure whether the measured rate is the same as the expected rate. We have used this tool to find out many issues in ChromeOS. I hope it can help development in Linux too. If you have any problems or want more features, please create a bug in crbug.com and assign to me ( yuhsuan(a)chromium.org). Thanks! Best, Yu-Hsuan

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