September 2020 - Alsa-devel - mailman.alsa-project.org

[RFC PATCH 1/2] ASoC: cs47l15: Fix EPOUT->HPOUT1 Mono Mux routing
by Richard Fitzgerald 25 Sep '20

25 Sep '20

EPOUT is always mono so should have a permanent routing through the HPOUT1 Mono Mux. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs47l15.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index a591e7457d11..254f9d96e766 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -1089,6 +1089,7 @@ static const struct snd_soc_dapm_route cs47l15_dapm_routes[] = { { "HPOUT1 Demux", NULL, "OUT1R" }, { "OUT1R", NULL, "HPOUT1 Mono Mux" }, + { "HPOUT1 Mono Mux", "EPOUT", "OUT1L" }, { "HPOUTL", "HPOUT", "HPOUT1 Demux" }, { "HPOUTR", "HPOUT", "HPOUT1 Demux" }, @@ -1268,7 +1269,6 @@ static irqreturn_t cs47l15_adsp2_irq(int irq, void *data) static const struct snd_soc_dapm_route cs47l15_mono_routes[] = { { "HPOUT1 Mono Mux", "HPOUT", "OUT1L" }, - { "HPOUT1 Mono Mux", "EPOUT", "OUT1L" }, }; static int cs47l15_component_probe(struct snd_soc_component *component) -- 2.20.1

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[PATCH v2] ALSA: hda - remove kerneldoc for internal hdac_i915 function
by Kai Vehmanen 24 Sep '20

24 Sep '20

Drop the kerneldoc markup for connectivity_check() as it's an static helper function. Fixes the following make W=1 warning: sound/hda/hdac_i915.c:80: warning: Function parameter or member 'i915' not described in 'connectivity_check' sound/hda/hdac_i915.c:80: warning: Function parameter or member 'hdac' not described in 'connectivity_check' Fixes: 7b882fe3e3e8 ('ALSA: hda - handle multiple i915 device instances') Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_i915.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index d236e497435d..5f0a1aa6ad84 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -73,9 +73,7 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus) } EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); -/** - * Returns true if the devices can be connected for audio. - */ +/* returns true if the devices can be connected for audio */ static bool connectivity_check(struct pci_dev *i915, struct pci_dev *hdac) { struct pci_bus *bus_a = i915->bus, *bus_b = hdac->bus; -- 2.27.0

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[PATCH] ALSA: hda - add missing documentation to hdac_i915
by Kai Vehmanen 24 Sep '20

24 Sep '20

Document missing function arguments and fix make W=1 warning: sound/hda/hdac_i915.c:80: warning: Function parameter or member 'i915' not described in 'connectivity_check' sound/hda/hdac_i915.c:80: warning: Function parameter or member 'hdac' not described in 'connectivity_check' Fixes: 7b882fe3e3e8 ('ALSA: hda - handle multiple i915 device instances') Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_i915.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index d236e497435d..9592d0c0c254 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -75,6 +75,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk); /** * Returns true if the devices can be connected for audio. + * @i915: i915 PCI device + * @hdac: HDA audio controller PCI device */ static bool connectivity_check(struct pci_dev *i915, struct pci_dev *hdac) { -- 2.27.0

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[PATCH] sound/soc/codecs: fix spelling typo in comments
by Wang Qing 24 Sep '20

24 Sep '20

Modify the comment typo: "compliment" -> "complement". Signed-off-by: Wang Qing <wangqing(a)vivo.com> --- sound/soc/codecs/ak4458.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4458.h b/sound/soc/codecs/ak4458.h index f906215..e43144c --- a/sound/soc/codecs/ak4458.h +++ b/sound/soc/codecs/ak4458.h @@ -49,7 +49,7 @@ /* DIF2 1 0 * x 1 0 MSB justified Figure 3 (default) - * x 1 1 I2S Compliment Figure 4 + * x 1 1 I2S Complement Figure 4 */ #define AK4458_DIF_SHIFT 1 #define AK4458_DIF_MASK GENMASK(3, 1) -- 2.7.4

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support the Roland UA-4FX2 card
by GitHub issues - opened 24 Sep '20

24 Sep '20

alsa-project/alsa-lib issue #81 was opened from kalashnikov2: hi i'm trying to use this audio card with a Raspberry pi and y get this from the command dmesg on linux console ``` usb 3-2: new high-speed USB device number 6 using xhci_hcd usb 3-2: New USB device found, idVendor=0582, idProduct=01e2, bcdDevice= 0.2b usb 3-2: New USB device strings: Mfr=1, Product=2, SerialNumber=0 usb 3-2: Product: UA-4FX2 usb 3-2: Manufacturer: ROLAND mc: Linux media interface: v0.10 usbcore: registered new interface driver snd-usb-audio usb 3-2: Unable to change format on ep #8e: already in use usb 3-2: Unable to change format on ep #8e: already in use : : ``` and when i try to select the device on a program get this ``` ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2642:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm_route.c:869:(find_matching_chmap) Found no matching channel map ALSA lib pcm_oss.c:377:(_snd_pcm_oss_open) Unknown field port ALSA lib pcm_oss.c:377:(_snd_pcm_oss_open) Unknown field port ALSA lib pcm_usb_stream.c:486:(_snd_pcm_usb_stream_open) Invalid type for card ALSA lib pcm_usb_stream.c:486:(_snd_pcm_usb_stream_open) Invalid type for card ``` it seems that the card is not supported by ALSA lib. How is the procedure to add new cards to the driver? How can I help to do this? Thank you Issue URL : https://github.com/alsa-project/alsa-lib/issues/81 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/5] ASoC: SOF: fix kcontrol size checks
by Kai Vehmanen 24 Sep '20

24 Sep '20

Series that fixes checks for 'size' in kcontrol get/put ext_bytes methods for SOF. The gaps in these checks were discovered via cppcheck warnings on unused variable values. Pierre-Louis Bossart (5): ASoC: SOF: control: fix size checks for ext_bytes control .get() ASoC: SOF: control: fix size checks for volatile ext_bytes control .get() ASoC: SOF: control: add size checks for ext_bytes control .put() ASoC: SOF: control: remove const in sizeof() ASoC: SOF: topology: remove const in sizeof() sound/soc/sof/control.c | 53 +++++++++++++++++++++++++++++++--------- sound/soc/sof/topology.c | 2 +- 2 files changed, 43 insertions(+), 12 deletions(-) -- 2.27.0

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No ack() calback in component driver
by Sameer Pujar 24 Sep '20

24 Sep '20

Hi Morimoto-san, With your commit e2cb4a14541d ("ASoC: soc-core: merge snd_pcm_ops member to component driver") snd_pcm_ops is replaced with callback functions in component driver. In doing so, ack() callback was not retained in component driver. Was this intentional because there were no users of ack() or if there were other reasons? Thanks, Sameer.

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Re: [greybus-dev] [PATCH 3/3] [PATCH] staging: greybus: __u8 is sufficient for snd_ctl_elem_type_t and snd_ctl_elem_iface_t
by Alex Elder 24 Sep '20

24 Sep '20

On 9/24/20 5:20 AM, Coiby Xu wrote: > Use __8 to replace int and remove the unnecessary __bitwise type attribute. > > Found by sparse, Greg's right, don't change the public header. You could try this in the Greybus code to eliminate the warning, but I haven't tried it (and for all I know it's not a good idea): uinfo->type = (__force snd_ctl_elem_type_t)info->type; -Alex > $ make C=2 drivers/staging/greybus/ > drivers/staging/greybus/audio_topology.c:185:24: warning: cast to restricted snd_ctl_elem_type_t > drivers/staging/greybus/audio_topology.c:679:14: warning: restricted snd_ctl_elem_iface_t degrades to integer > drivers/staging/greybus/audio_topology.c:906:14: warning: restricted snd_ctl_elem_iface_t degrades to integer > > Signed-off-by: Coiby Xu <coiby.xu(a)gmail.com> > --- > drivers/staging/greybus/audio_topology.c | 2 +- > include/uapi/sound/asound.h | 4 ++-- > 2 files changed, 3 insertions(+), 3 deletions(-) > > diff --git a/drivers/staging/greybus/audio_topology.c b/drivers/staging/greybus/audio_topology.c > index 56bf1a4f95ad..f6023ff390c2 100644 > --- a/drivers/staging/greybus/audio_topology.c > +++ b/drivers/staging/greybus/audio_topology.c > @@ -182,7 +182,7 @@ static int gbcodec_mixer_ctl_info(struct snd_kcontrol *kcontrol, > /* update uinfo */ > uinfo->access = data->access; > uinfo->count = data->vcount; > - uinfo->type = (snd_ctl_elem_type_t)info->type; > + uinfo->type = info->type; > > switch (info->type) { > case GB_AUDIO_CTL_ELEM_TYPE_BOOLEAN: > diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h > index 535a7229e1d9..8e71a95644ab 100644 > --- a/include/uapi/sound/asound.h > +++ b/include/uapi/sound/asound.h > @@ -950,7 +950,7 @@ struct snd_ctl_card_info { > unsigned char components[128]; /* card components / fine identification, delimited with one space (AC97 etc..) */ > }; > > -typedef int __bitwise snd_ctl_elem_type_t; > +typedef __u8 snd_ctl_elem_type_t; > #define SNDRV_CTL_ELEM_TYPE_NONE ((__force snd_ctl_elem_type_t) 0) /* invalid */ > #define SNDRV_CTL_ELEM_TYPE_BOOLEAN ((__force snd_ctl_elem_type_t) 1) /* boolean type */ > #define SNDRV_CTL_ELEM_TYPE_INTEGER ((__force snd_ctl_elem_type_t) 2) /* integer type */ > @@ -960,7 +960,7 @@ typedef int __bitwise snd_ctl_elem_type_t; > #define SNDRV_CTL_ELEM_TYPE_INTEGER64 ((__force snd_ctl_elem_type_t) 6) /* 64-bit integer type */ > #define SNDRV_CTL_ELEM_TYPE_LAST SNDRV_CTL_ELEM_TYPE_INTEGER64 > > -typedef int __bitwise snd_ctl_elem_iface_t; > +typedef __u8 snd_ctl_elem_iface_t; > #define SNDRV_CTL_ELEM_IFACE_CARD ((__force snd_ctl_elem_iface_t) 0) /* global control */ > #define SNDRV_CTL_ELEM_IFACE_HWDEP ((__force snd_ctl_elem_iface_t) 1) /* hardware dependent device */ > #define SNDRV_CTL_ELEM_IFACE_MIXER ((__force snd_ctl_elem_iface_t) 2) /* virtual mixer device */ >

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Re: [PATCH 3/3] [PATCH] staging: greybus: __u8 is sufficient for snd_ctl_elem_type_t and snd_ctl_elem_iface_t
by Greg Kroah-Hartman 24 Sep '20

24 Sep '20

On Thu, Sep 24, 2020 at 06:20:39PM +0800, Coiby Xu wrote: > Use __8 to replace int and remove the unnecessary __bitwise type attribute. > > Found by sparse, > > $ make C=2 drivers/staging/greybus/ > drivers/staging/greybus/audio_topology.c:185:24: warning: cast to restricted snd_ctl_elem_type_t > drivers/staging/greybus/audio_topology.c:679:14: warning: restricted snd_ctl_elem_iface_t degrades to integer > drivers/staging/greybus/audio_topology.c:906:14: warning: restricted snd_ctl_elem_iface_t degrades to integer > > Signed-off-by: Coiby Xu <coiby.xu(a)gmail.com> > --- > drivers/staging/greybus/audio_topology.c | 2 +- > include/uapi/sound/asound.h | 4 ++-- > 2 files changed, 3 insertions(+), 3 deletions(-) > > diff --git a/drivers/staging/greybus/audio_topology.c b/drivers/staging/greybus/audio_topology.c > index 56bf1a4f95ad..f6023ff390c2 100644 > --- a/drivers/staging/greybus/audio_topology.c > +++ b/drivers/staging/greybus/audio_topology.c > @@ -182,7 +182,7 @@ static int gbcodec_mixer_ctl_info(struct snd_kcontrol *kcontrol, > /* update uinfo */ > uinfo->access = data->access; > uinfo->count = data->vcount; > - uinfo->type = (snd_ctl_elem_type_t)info->type; > + uinfo->type = info->type; > > switch (info->type) { > case GB_AUDIO_CTL_ELEM_TYPE_BOOLEAN: > diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h > index 535a7229e1d9..8e71a95644ab 100644 > --- a/include/uapi/sound/asound.h > +++ b/include/uapi/sound/asound.h > @@ -950,7 +950,7 @@ struct snd_ctl_card_info { > unsigned char components[128]; /* card components / fine identification, delimited with one space (AC97 etc..) */ > }; > > -typedef int __bitwise snd_ctl_elem_type_t; > +typedef __u8 snd_ctl_elem_type_t; > #define SNDRV_CTL_ELEM_TYPE_NONE ((__force snd_ctl_elem_type_t) 0) /* invalid */ > #define SNDRV_CTL_ELEM_TYPE_BOOLEAN ((__force snd_ctl_elem_type_t) 1) /* boolean type */ > #define SNDRV_CTL_ELEM_TYPE_INTEGER ((__force snd_ctl_elem_type_t) 2) /* integer type */ > @@ -960,7 +960,7 @@ typedef int __bitwise snd_ctl_elem_type_t; > #define SNDRV_CTL_ELEM_TYPE_INTEGER64 ((__force snd_ctl_elem_type_t) 6) /* 64-bit integer type */ > #define SNDRV_CTL_ELEM_TYPE_LAST SNDRV_CTL_ELEM_TYPE_INTEGER64 > > -typedef int __bitwise snd_ctl_elem_iface_t; > +typedef __u8 snd_ctl_elem_iface_t; > #define SNDRV_CTL_ELEM_IFACE_CARD ((__force snd_ctl_elem_iface_t) 0) /* global control */ > #define SNDRV_CTL_ELEM_IFACE_HWDEP ((__force snd_ctl_elem_iface_t) 1) /* hardware dependent device */ > #define SNDRV_CTL_ELEM_IFACE_MIXER ((__force snd_ctl_elem_iface_t) 2) /* virtual mixer device */ I can't take a uapi sound header file patch along with a driver change, these need to be independant. And this is a userspace api, you just changed the size of an externally facing variable, are you _SURE_ that is ok to do? thanks, greg k-h

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RE: [PATCH 3/3] [PATCH] staging: greybus: __u8 is sufficient for snd_ctl_elem_type_t and snd_ctl_elem_iface_t
by David Laight 24 Sep '20

24 Sep '20

From: Coiby Xu > Sent: 24 September 2020 11:21 > Use __8 to replace int and remove the unnecessary __bitwise type attribute. > > Found by sparse, ... > diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h > index 535a7229e1d9..8e71a95644ab 100644 > --- a/include/uapi/sound/asound.h > +++ b/include/uapi/sound/asound.h > @@ -950,7 +950,7 @@ struct snd_ctl_card_info { > unsigned char components[128]; /* card components / fine identification, delimited with one > space (AC97 etc..) */ > }; > > -typedef int __bitwise snd_ctl_elem_type_t; > +typedef __u8 snd_ctl_elem_type_t; > #define SNDRV_CTL_ELEM_TYPE_NONE ((__force snd_ctl_elem_type_t) 0) /* invalid */ > #define SNDRV_CTL_ELEM_TYPE_BOOLEAN ((__force snd_ctl_elem_type_t) 1) /* boolean type */ > #define SNDRV_CTL_ELEM_TYPE_INTEGER ((__force snd_ctl_elem_type_t) 2) /* integer type */ WTF is all that about anyway?? What is wrong with: #define SNDRV_CTL_ELEM_TYPE_NONE 0u /* invalid */ David - Registered Address Lakeside, Bramley Road, Mount Farm, Milton Keynes, MK1 1PT, UK Registration No: 1397386 (Wales)

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[PATCH v6 0/5] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 23 Sep '20

23 Sep '20

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 changes since V5: -- Removed unused struct regmap *map in lpass_platform_alloc_hdmidmactl_fields. -- DMA alloc and free API signature change in lpass-apq8016.c, lpass-ipq806x.c -- Keeping API "irqreturn_t lpass_platform_hdmiif_irq" under ifdef macro Changes Since v4: -- Updated with single compatible node for both I2S and HDMI. Changes Since v3: -- Removed id in lpass variant structure and used snd_soc_dai_driver id. Changes Since v2: -- Audio buffer size(i.e. LPASS_PLATFORM_BUFFER_SIZE) in lpass-platform.c increased. Changes Since v1: -- Commit messages are updated -- Addressed Rob Herring review comments V Sujith Kumar Reddy (5): ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi Asoc:qcom:lpass-cpu:Update dts property read API ASoC: qcom: Add support for lpass hdmi driver ASoC: qcom: sc7180: Add support for audio over DP .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 74 ++- include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 4 +- sound/soc/qcom/lpass-cpu.c | 39 +- sound/soc/qcom/lpass-hdmi.c | 596 +++++++++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 129 +++++ sound/soc/qcom/lpass-ipq806x.c | 4 +- sound/soc/qcom/lpass-lpaif-reg.h | 52 +- sound/soc/qcom/lpass-platform.c | 468 ++++++++++++---- sound/soc/qcom/lpass-sc7180.c | 116 +++- sound/soc/qcom/lpass.h | 119 +++- 13 files changed, 1451 insertions(+), 158 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH 0/6] ASoC: Intel: sdw machine driver updates for 5.10
by Kai Vehmanen 23 Sep '20

23 Sep '20

Series including fixes and improvements for Intel SoundWire machine drivers. Bard Liao (1): ASoC: Intel: add support for new SoundWire hardware layout on TGL Pierre-Louis Bossart (4): ASoC: Intel: sof_sdw: remove ternary operator ASoC: Intel: add codec name prefix to ACPI machine description ASoC: Intel: sof_sdw: remove hard-coded codec_conf table ASoC: Intel: sof_sdw_rt700: add codec prefix Rander Wang (1): ASOC: Intel: sof_sdw: restore playback functionality with max98373 amps include/sound/soc-acpi.h | 2 + sound/soc/intel/boards/sof_sdw.c | 170 +++++++++--------- sound/soc/intel/boards/sof_sdw_common.h | 3 + sound/soc/intel/boards/sof_sdw_max98373.c | 36 +++- sound/soc/intel/boards/sof_sdw_rt700.c | 6 +- .../intel/common/soc-acpi-intel-cml-match.c | 10 ++ .../intel/common/soc-acpi-intel-cnl-match.c | 1 + .../intel/common/soc-acpi-intel-icl-match.c | 6 + .../intel/common/soc-acpi-intel-tgl-match.c | 67 +++++++ 9 files changed, 216 insertions(+), 85 deletions(-) -- 2.27.0

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[PATCH 1/6] dt-bindings: tas2770: Add shutdown gpio property
by Dan Murphy 23 Sep '20

23 Sep '20

Add the shutdown-gpios property to the yaml to define the GPIO that can be used to place the device in shutdown mode or wake the device up. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tas2770.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index bb26d081c9fa..9fdf614add55 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -29,6 +29,9 @@ properties: reset-gpio: description: GPIO used to reset the device. + shutdown-gpios: + description: GPIO used to control the state of the device. + interrupts: maxItems: 1 @@ -69,6 +72,7 @@ examples: interrupt-parent = <&gpio1>; interrupts = <14>; reset-gpio = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 14 0>; ti,imon-slot-no = <0>; ti,vmon-slot-no = <2>; }; -- 2.28.0

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[PATCH] ASoC: Intel: hda_dsp_common: use static function in conditional block
by Kai Vehmanen 23 Sep '20

23 Sep '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> cppcheck reports the following warning: sound/soc/intel/boards/hda_dsp_common.c:17:0: style: The function 'hda_dsp_hdmi_pcm_handle' is never used. [unusedFunction] Fix by moving to static inside compilation block. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/intel/boards/hda_dsp_common.c | 7 ++++--- sound/soc/intel/boards/hda_dsp_common.h | 3 --- 2 files changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/hda_dsp_common.c b/sound/soc/intel/boards/hda_dsp_common.c index 244b57fba64c..91ad2a0ad1ce 100644 --- a/sound/soc/intel/boards/hda_dsp_common.c +++ b/sound/soc/intel/boards/hda_dsp_common.c @@ -10,12 +10,14 @@ #include "hda_dsp_common.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) + /* * Search card topology and return PCM device number * matching Nth HDMI device (zero-based index). */ -struct snd_pcm *hda_dsp_hdmi_pcm_handle(struct snd_soc_card *card, - int hdmi_idx) +static struct snd_pcm *hda_dsp_hdmi_pcm_handle(struct snd_soc_card *card, + int hdmi_idx) { struct snd_soc_pcm_runtime *rtd; struct snd_pcm *spcm; @@ -34,7 +36,6 @@ struct snd_pcm *hda_dsp_hdmi_pcm_handle(struct snd_soc_card *card, return NULL; } -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) /* * Search card topology and register HDMI PCM related controls * to codec driver. diff --git a/sound/soc/intel/boards/hda_dsp_common.h b/sound/soc/intel/boards/hda_dsp_common.h index 727edd256962..ea4ae9285cf0 100644 --- a/sound/soc/intel/boards/hda_dsp_common.h +++ b/sound/soc/intel/boards/hda_dsp_common.h @@ -15,9 +15,6 @@ #include <sound/hda_i915.h> #include "../../codecs/hdac_hda.h" -struct snd_pcm *hda_dsp_hdmi_pcm_handle(struct snd_soc_card *card, - int hdmi_idx); - #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) int hda_dsp_hdmi_build_controls(struct snd_soc_card *card, struct snd_soc_component *comp); -- 2.27.0

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[PATCH v2 0/6] ASoC: Intel: sdw machine driver updates for 5.10
by Kai Vehmanen 23 Sep '20

23 Sep '20

Series including fixes and improvements for Intel SoundWire machine drivers. v2 changes: - fix the incorrct commit ids in Fixes tags Bard Liao (1): ASoC: Intel: add support for new SoundWire hardware layout on TGL Pierre-Louis Bossart (4): ASoC: Intel: sof_sdw: remove ternary operator ASoC: Intel: add codec name prefix to ACPI machine description ASoC: Intel: sof_sdw: remove hard-coded codec_conf table ASoC: Intel: sof_sdw_rt700: add codec prefix Rander Wang (1): ASOC: Intel: sof_sdw: restore playback functionality with max98373 amps include/sound/soc-acpi.h | 2 + sound/soc/intel/boards/sof_sdw.c | 170 +++++++++--------- sound/soc/intel/boards/sof_sdw_common.h | 3 + sound/soc/intel/boards/sof_sdw_max98373.c | 36 +++- sound/soc/intel/boards/sof_sdw_rt700.c | 6 +- .../intel/common/soc-acpi-intel-cml-match.c | 10 ++ .../intel/common/soc-acpi-intel-cnl-match.c | 1 + .../intel/common/soc-acpi-intel-icl-match.c | 6 + .../intel/common/soc-acpi-intel-tgl-match.c | 67 +++++++ 9 files changed, 216 insertions(+), 85 deletions(-) -- 2.27.0

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[PATCH] ALSA: usb-audio: Add mixer support for Pioneer DJ DJM-250MK2
by František Kučera 23 Sep '20

23 Sep '20

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows mapping playback and capture channels to available sources. Configures the card through USB commands. Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/mixer_quirks.c | 213 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 213 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199cdbfdc761..df036a359f2f 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2602,6 +2602,216 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * Pioneer DJ DJM-250MK2 and maybe other DJM models + * + * For playback, no duplicate mapping should be set. + * There are three mixer stereo channels (CH1, CH2, AUX) + * and three stereo sources (Playback 1-2, Playback 3-4, Playback 5-6). + * Each channel should be mapped just once to one source. + * If mapped multiple times, only one source will play on given channel + * (sources are not mixed together). + * + * For recording, duplicate mapping is OK. We will get the same signal multiple times. + * + * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN. + * + * See also notes in the quirks-table.h file. + */ + +struct snd_pioneer_djm_option { + const u16 wIndex; + const u16 wValue; + const char *name; +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_level[] = { + { .name = "-5 dB", .wValue = 0x0300, .wIndex = 0x8003 }, + { .name = "-10 dB", .wValue = 0x0200, .wIndex = 0x8003 }, + { .name = "-15 dB", .wValue = 0x0100, .wIndex = 0x8003 }, + { .name = "-19 dB", .wValue = 0x0000, .wIndex = 0x8003 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch12[] = { + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0109, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x010d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x010a, .wIndex = 0x8002 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch34[] = { + { .name = "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0209, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x020d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x020a, .wIndex = 0x8002 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch56[] = { + { .name = "REC OUT", .wValue = 0x030a, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0311, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0312, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0309, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x030d, .wIndex = 0x8002 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_12[] = { + { .name = "CH1", .wValue = 0x0100, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0101, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0104, .wIndex = 0x8016 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_34[] = { + { .name = "CH1", .wValue = 0x0200, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0201, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0204, .wIndex = 0x8016 } +}; + +static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_56[] = { + { .name = "CH1", .wValue = 0x0300, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0301, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0304, .wIndex = 0x8016 } +}; + +struct snd_pioneer_djm_option_group { + const char *name; + const struct snd_pioneer_djm_option *options; + const size_t count; + const u16 default_value; +}; + +#define snd_pioneer_djm_option_group_item(_name, suffix, _default_value) { \ + .name = _name, \ + .options = snd_pioneer_djm_options_##suffix, \ + .count = ARRAY_SIZE(snd_pioneer_djm_options_##suffix), \ + .default_value = _default_value } + +static const struct snd_pioneer_djm_option_group snd_pioneer_djm_option_groups[] = { + snd_pioneer_djm_option_group_item("Master Capture Level Capture Switch", capture_level, 0), + snd_pioneer_djm_option_group_item("Capture 1-2 Capture Switch", capture_ch12, 2), + snd_pioneer_djm_option_group_item("Capture 3-4 Capture Switch", capture_ch34, 2), + snd_pioneer_djm_option_group_item("Capture 5-6 Capture Switch", capture_ch56, 0), + snd_pioneer_djm_option_group_item("Playback 1-2 Playback Switch", playback_12, 0), + snd_pioneer_djm_option_group_item("Playback 3-4 Playback Switch", playback_34, 1), + snd_pioneer_djm_option_group_item("Playback 5-6 Playback Switch", playback_56, 2) +}; + +// layout of the kcontrol->private_value: +#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff +#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000 +#define SND_PIONEER_DJM_GROUP_SHIFT 16 + +static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *info) +{ + u16 group_index = kctl->private_value >> SND_PIONEER_DJM_GROUP_SHIFT; + size_t count; + const char *name; + const struct snd_pioneer_djm_option_group *group; + + if (group_index >= ARRAY_SIZE(snd_pioneer_djm_option_groups)) + return -EINVAL; + + group = &snd_pioneer_djm_option_groups[group_index]; + count = group->count; + if (info->value.enumerated.item >= count) + info->value.enumerated.item = count - 1; + name = group->options[info->value.enumerated.item].name; + strlcpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name)); + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = count; + return 0; +} + +static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value) +{ + int err; + + if (group >= ARRAY_SIZE(snd_pioneer_djm_option_groups) + || value >= snd_pioneer_djm_option_groups[group].count) + return -EINVAL; + + err = snd_usb_lock_shutdown(mixer->chip); + if (err) + return err; + + err = snd_usb_ctl_msg( + mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), + USB_REQ_SET_FEATURE, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + snd_pioneer_djm_option_groups[group].options[value].wValue, + snd_pioneer_djm_option_groups[group].options[value].wIndex, + NULL, 0); + + snd_usb_unlock_shutdown(mixer->chip); + return err; +} + +static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + elem->value.enumerated.item[0] = kctl->private_value & SND_PIONEER_DJM_VALUE_MASK; + return 0; +} + +static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + unsigned long private_value = kctl->private_value; + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = elem->value.enumerated.item[0]; + + kctl->private_value = (group << SND_PIONEER_DJM_GROUP_SHIFT) | value; + + return snd_pioneer_djm_controls_update(mixer, group, value); +} + +static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list) +{ + unsigned long private_value = list->kctl->private_value; + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = (private_value & SND_PIONEER_DJM_VALUE_MASK); + + return snd_pioneer_djm_controls_update(list->mixer, group, value); +} + +static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + const struct snd_pioneer_djm_option_group *group; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_pioneer_djm_controls_info, + .get = snd_pioneer_djm_controls_get, + .put = snd_pioneer_djm_controls_put + }; + + for (i = 0; i < ARRAY_SIZE(snd_pioneer_djm_option_groups); i++) { + group = &snd_pioneer_djm_option_groups[i]; + knew.name = group->name; + knew.private_value = (i << SND_PIONEER_DJM_GROUP_SHIFT) | group->default_value; + err = snd_pioneer_djm_controls_update(mixer, i, group->default_value); + if (err) + return err; + err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume, + &knew, NULL); + if (err) + return err; + } + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2706,6 +2916,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ + err = snd_pioneer_djm_controls_create(mixer); + break; } return err; -- 2.20.1

2 1

[PATCH] soundwire: remove an unnecessary NULL check
by Dan Carpenter 23 Sep '20

23 Sep '20

The "bus" pointer isn't NULL so the address to a non-zero offset in middle of "bus" cannot be NULL. Delete the NULL check. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- drivers/soundwire/generic_bandwidth_allocation.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/drivers/soundwire/generic_bandwidth_allocation.c b/drivers/soundwire/generic_bandwidth_allocation.c index 6088775b67a5..fdc0db71360e 100644 --- a/drivers/soundwire/generic_bandwidth_allocation.c +++ b/drivers/soundwire/generic_bandwidth_allocation.c @@ -342,15 +342,11 @@ static int sdw_select_row_col(struct sdw_bus *bus, int clk_freq) static int sdw_compute_bus_params(struct sdw_bus *bus) { unsigned int max_dr_freq, curr_dr_freq = 0; - struct sdw_master_prop *mstr_prop = NULL; + struct sdw_master_prop *mstr_prop = &bus->prop; int i, clk_values, ret; bool is_gear = false; u32 *clk_buf; - mstr_prop = &bus->prop; - if (!mstr_prop) - return -EINVAL; - if (mstr_prop->num_clk_gears) { clk_values = mstr_prop->num_clk_gears; clk_buf = mstr_prop->clk_gears; -- 2.28.0

2 1

[PATCH v2 0/2] soundwire: sysfs: expose device number and status
by Pierre-Louis Bossart 23 Sep '20

23 Sep '20

This patchset combines three contributions: Srinivas Kandagalta suggested creating a device even it's detected on the bus but not described in platform firmware. I suggested adding the device number and status to show the difference with 'ghost' devices, described in firmware but not physically present. Vinod suggested a simpler way to report the status. I did not keep Vinod's patch separate since it was using the same group attribute as the other properties, which prevents status and device number from being reported if there is no firmware and no driver. These patches were tested on Qualcomm and Intel platforms. v2: as suggested by GregKH, add attribute group by default by setting the groups pointer at the device level. Pierre-Louis Bossart (1): soundwire: sysfs: add slave status and device number before probe Srinivas Kandagatla (1): soundwire: bus: add enumerated Slave device to device list .../ABI/testing/sysfs-bus-soundwire-slave | 18 ++++++ drivers/soundwire/bus.c | 9 +++ drivers/soundwire/bus.h | 2 + drivers/soundwire/bus_type.c | 9 +++ drivers/soundwire/slave.c | 6 +- drivers/soundwire/sysfs_local.h | 4 ++ drivers/soundwire/sysfs_slave.c | 59 ++++++++++++++++++- 7 files changed, 104 insertions(+), 3 deletions(-) -- 2.25.1

3 11

[PATCH 0/3] soundwire: enable Data Port test modes
by Bard Liao 23 Sep '20

23 Sep '20

Test modes are required for all SoundWire IP, and help debug integration issues. This series adds debugfs support and data port test fail interrupt to enable data port test mode feature on Intel platforms. Pierre-Louis Bossart (3): soundwire: enable Data Port test modes soundwire: intel: enable test modes soundwire: cadence: add data port test fail interrupt drivers/soundwire/bus.c | 6 +++ drivers/soundwire/cadence_master.c | 27 ++++++++++-- .../soundwire/generic_bandwidth_allocation.c | 6 ++- drivers/soundwire/intel.c | 42 +++++++++++++++++++ drivers/soundwire/stream.c | 3 +- include/linux/soundwire/sdw.h | 6 +++ 6 files changed, 84 insertions(+), 6 deletions(-) -- 2.17.1

2 4

[PATCH AUTOSEL 5.8 03/20] ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811
by Sasha Levin 22 Sep '20

22 Sep '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit 811c5494436789e7149487c06e0602b507ce274b ] The WM8994_MICBIAS register is not available in the WM1811 CODEC so skip initialization of that register for that device. This suppresses an error during boot: "wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec" Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200827173357.31891-1-s.nawrocki@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm_hubs.c | 3 +++ sound/soc/codecs/wm_hubs.h | 1 + 3 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 55d0b9be6ff00..c2116836a7203 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4166,11 +4166,13 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994->hubs.dcs_readback_mode = 2; break; } + wm8994->hubs.micd_scthr = true; break; case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.micd_scthr = true; switch (control->revision) { case 0: diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e93af7edd8f75..dd421e2fe7b21 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1223,6 +1223,9 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + if (!hubs->micd_scthr) + return 0; + snd_soc_component_update_bits(component, WM8993_MICBIAS, WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | WM8993_MICB1_LVL | WM8993_MICB2_LVL, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 4b8e5f0d6e32d..988b29e630607 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -27,6 +27,7 @@ struct wm_hubs_data { int hp_startup_mode; int series_startup; int no_series_update; + bool micd_scthr; bool no_cache_dac_hp_direct; struct list_head dcs_cache; -- 2.25.1

2 4

[PATCH v3 1/1] ALSA: hda: Refactor calculating SDnFMT according to specification
by Pawel Harlozinski 22 Sep '20

22 Sep '20

Fix setting SDnFMT based on :wq High Definition Audio Specification Rev. 1.0a page 48. Bits per Sample (BITS): 000 = 8 bits. The data will be packed in memory in 8-bit containers on 16-bit boundaries. 001 = 16 bits. The data will be packed in memory in 16-bit containers on 16-bit boundaries. 010 = 20 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 011 = 24 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 100 = 32 bits. The data will be packed in memory in 32-bit containers on 32-bit boundaries. 101-111 = Reserved Set SDnFMT depending on which format was given. Henceforth split cases for formats 20, 24, 32 bits, but leave constraints to maxbps. Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- v3: drop gerrit Change-Id v2: leave constraints to maxbps sound/hda/hdac_device.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 3e9e9ac804f6..ccc47a10ba63 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -764,7 +764,14 @@ unsigned int snd_hdac_calc_stream_format(unsigned int rate, val |= AC_FMT_BITS_16; break; case 20: + val |= AC_FMT_BITS_20; + break; case 24: + if (maxbps >= 24) + val |= AC_FMT_BITS_24; + else + val |= AC_FMT_BITS_20; + break; case 32: if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE) val |= AC_FMT_BITS_32; -- 2.17.1

2 1

[PATCH v2 00/13] gpio: add common dtschema
by Krzysztof Kozlowski 22 Sep '20

22 Sep '20

Hi, Changes since v1 ================ 1. Use common schema in bindings outside of gpio/ (new patches). 2. Minor fixes - see individual patches for changelogs. Notes ===== This is independent work of pca953x bindings: https://lore.kernel.org/lkml/20200916155715.21009-1-krzk@kernel.org/T/#u The DTS patches can be also applied independently. The bindings patches depend on first one, adding common schema. Best regards, Krzysztof Krzysztof Kozlowski (13): dt-bindings: gpio: add common schema for GPIO controllers dt-bindings: gpio: include common schema in GPIO controllers dt-bindings: gpio: pl061: add missing properties and include common schema dt-bindings: gpio: fsl-imx-gpio: add i.MX ARMv6 and ARMv7 compatibles dt-bindings: gpio: fsl-imx-gpio: add gpio-line-names dt-bindings: gpio: gpio-vf610: fix iMX 7ULP compatible matching dt-bindings: media: include common schema in GPIO controllers dt-bindings: mfd: include common schema in GPIO controllers dt-bindings: pinctrl: include common schema in GPIO controllers ASoC: dt-bindings: zl38060: include common schema in GPIO controllers arm64: dts: imx8mq-librem5: correct GPIO hog property arm64: dts: imx8mq-librem5: align GPIO hog names with dtschema ARM: dts: imx: align GPIO hog names with dtschema .../bindings/gpio/brcm,xgs-iproc-gpio.yaml | 3 + .../bindings/gpio/fsl-imx-gpio.yaml | 17 ++- .../devicetree/bindings/gpio/gpio-common.yaml | 125 ++++++++++++++++++ .../devicetree/bindings/gpio/gpio-mxs.yaml | 4 + .../bindings/gpio/gpio-pca9570.yaml | 3 + .../devicetree/bindings/gpio/gpio-rda.yaml | 3 + .../devicetree/bindings/gpio/gpio-vf610.yaml | 11 +- .../devicetree/bindings/gpio/mrvl-gpio.yaml | 1 + .../devicetree/bindings/gpio/pl061-gpio.yaml | 6 + .../bindings/gpio/qcom,wcd934x-gpio.yaml | 3 + .../bindings/gpio/renesas,em-gio.yaml | 3 + .../bindings/gpio/renesas,rcar-gpio.yaml | 3 + .../devicetree/bindings/gpio/sifive,gpio.yaml | 3 + .../bindings/gpio/snps,dw-apb-gpio.yaml | 3 + .../gpio/socionext,uniphier-gpio.yaml | 3 + .../bindings/gpio/xylon,logicvc-gpio.yaml | 3 + .../bindings/media/i2c/maxim,max9286.yaml | 3 + .../bindings/mfd/cirrus,madera.yaml | 1 + .../devicetree/bindings/mfd/max77650.yaml | 3 + .../bindings/mfd/rohm,bd71828-pmic.yaml | 3 + .../devicetree/bindings/mfd/st,stmfx.yaml | 2 + .../devicetree/bindings/mfd/wlf,arizona.yaml | 1 + .../pinctrl/actions,s500-pinctrl.yaml | 3 + .../pinctrl/allwinner,sun4i-a10-pinctrl.yaml | 1 + .../bindings/pinctrl/cirrus,lochnagar.yaml | 3 + .../bindings/pinctrl/ingenic,pinctrl.yaml | 3 + .../pinctrl/mediatek,mt6779-pinctrl.yaml | 3 + .../bindings/pinctrl/pinctrl-mt8192.yaml | 3 + .../pinctrl/qcom,ipq6018-pinctrl.yaml | 3 + .../pinctrl/qcom,msm8226-pinctrl.yaml | 3 + .../bindings/pinctrl/qcom,sm8250-pinctrl.yaml | 3 + .../pinctrl/renesas,rza2-pinctrl.yaml | 3 + .../bindings/pinctrl/st,stm32-pinctrl.yaml | 3 + .../devicetree/bindings/sound/zl38060.yaml | 3 + arch/arm/boot/dts/imx51-zii-rdu1.dts | 2 +- arch/arm/boot/dts/imx6qdl-zii-rdu2.dtsi | 8 +- arch/arm/boot/dts/imx6ul-ccimx6ulsbcpro.dts | 2 +- .../boot/dts/freescale/imx8mq-librem5.dtsi | 4 +- 38 files changed, 245 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/gpio/gpio-common.yaml -- 2.17.1

7 25

[PATCH] Revert "ALSA: usb-audio: Disable Lenovo P620 Rear line-in volume control"
by Kai-Heng Feng 22 Sep '20

22 Sep '20

This reverts commit 34dedd2a83b241ba6aeb290260313c65dc58660e. According to Realtek, volume FU works for line-in. I can confirm volume control works after device firmware is updated. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/usb/mixer_maps.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 5b43e9e40e49..c369c81e74c4 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -371,7 +371,6 @@ static const struct usbmix_name_map asus_rog_map[] = { }; static const struct usbmix_name_map lenovo_p620_rear_map[] = { - { 19, NULL, 2 }, /* FU, Volume */ { 19, NULL, 12 }, /* FU, Input Gain Pad */ {} }; -- 2.17.1

2 3

[PATCH] ASoC: tas2562: Remove duplicate code for I/V sense
by Dan Murphy 22 Sep '20

22 Sep '20

Remove duplicate code for programming the I/V sense the call to update the register was duplicated in commit 09ed395b05feb ("ASoC: tas2562: Add voltage sense slot configuration"). Fixes: 09ed395b05feb ("ASoC: tas2562: Add voltage sense slot configuration") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 12 ------------ 1 file changed, 12 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index da820e8d59a1..f1ff204e3ad0 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -253,18 +253,6 @@ static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, if (ret < 0) return ret; - ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG5, - TAS2562_TDM_CFG5_VSNS_SLOT_MASK, - tas2562->v_sense_slot); - if (ret < 0) - return ret; - - ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG6, - TAS2562_TDM_CFG6_ISNS_SLOT_MASK, - tas2562->i_sense_slot); - if (ret < 0) - return ret; - return 0; } -- 2.28.0

2 1

[PATCH] ASoC: cros_ec_codec: fix kconfig dependency warning for SND_SOC_CROS_EC_CODEC
by Necip Fazil Yildiran 22 Sep '20

22 Sep '20

When SND_SOC_CROS_EC_CODEC is enabled and CRYPTO is disabled, it results in the following Kbuild warning: WARNING: unmet direct dependencies detected for CRYPTO_LIB_SHA256 Depends on [n]: CRYPTO [=n] Selected by [y]: - SND_SOC_CROS_EC_CODEC [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && CROS_EC [=y] The reason is that SND_SOC_CROS_EC_CODEC selects CRYPTO_LIB_SHA256 without depending on or selecting CRYPTO while CRYPTO_LIB_SHA256 is subordinate to CRYPTO. Honor the kconfig menu hierarchy to remove kconfig dependency warnings. Fixes: 93fa0af4790a ("ASoC: cros_ec_codec: switch to library API for SHA-256") Signed-off-by: Necip Fazil Yildiran <fazilyildiran(a)gmail.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 946a70210f49..601ea45d3ea6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -540,6 +540,7 @@ config SND_SOC_CQ0093VC config SND_SOC_CROS_EC_CODEC tristate "codec driver for ChromeOS EC" depends on CROS_EC + select CRYPTO select CRYPTO_LIB_SHA256 help If you say yes here you will get support for the -- 2.25.1

2 1

[PATCH v7 00/14] ASoC: Intel: Catpt - Lynx and Wildcat point
by Cezary Rojewski 22 Sep '20

22 Sep '20

Implement support for Lynxpoint and Wildcat Point AudioDSP. Catpt solution deprecates existing sound/soc/intel/haswell which is removed in the following series. Due to high range of errors and desynchronization from recommendations set by Windows solution, re-write came as a lower-cost solution compared to refactoring /haswell/ with several series of patches. Series is dependent on linux-spi change: spi: pxa2xx: Add SSC2 and SSPSP2 SSP registers https://www.spinics.net/lists/linux-spi/msg23885.html which has been already merged and is now part of linux-spi tree. Bulk of series content is device driver core code - everything up to patch 7/14 - with fs entries and trace macros introduced right after. While each core patch is shaped in such a way that no unavailable members are ever called, until patch 14/14 is applied, no code compilation can occur as no Makefile is present. Once said patch is added, Makefile and Kconfig are implemented and driver module compiles as expected. Special thanks go to Marcin Barlik and Piotr Papierkowski for sharing their LPT/WPT AudioDSP architecture expertise as well as helping backtrack its historical background. My thanks go to Amadeusz Slawinski for reviews and improvements proposed on and off the internal list. Most of internal diff below is his contribution. Krzysztof Hejmowski helped me setup my own Xtensa environment and recompile LPT/WPT FW binary sources what sped up the development greatly. This would not have been possible without help from these champions, especially considering how quickly the catpt was written: 2 weeks features, 3 weeks optimizations. Thank you. Userspace-exposed members are compatible with what is exposed by deprecated solution as well as FW binary being re-used thus no harm is done. The only visible differences are: the newly added 'Loopback Mute' kcontrol and volume support extending to quad from stereo. On top of fixing erros and design flows, catpt also adds module reload, dynamic SRAM memory allocation during PCM runtime and exposes missing userspace API: 'Loopback Mute' kcontrol, quad volume controls and sysfs fw-version entries. Event tracing is provided to easy solution debugging. Following are not included in this update and are scheduled as later addition: - fw logging - module (library) support Note: LPT power up/down sequences might get aligned with WPT once enough testing is done as capabilities are shared for both DSPs. Note #2: Both LPT and WPT power up/down sequences may get optimized in future updates as thanks to help from the Windows team, most of nuances behind why/what/when in regard to hw registers have been backtracked and reviewed again. Link to developer's deep dive message: https://www.spinics.net/lists/alsa-devel/msg113563.html Changes in v7: - fixed licence header for fs.c - renamed fs.c to sysfs.c to better match its purpose - added documentation within Documentation/ABI/testing for entries exposed by catpt - bin_attribute fw_build replaced by attribute fw_info: fw_info contains full FW information and after successful handshake, it's always available (stored in driver data) so no need to invoke GET_FW_VERSION IPC again, just dump the stored information - rather than manually creating and removing sysfs files, now makes use of dev_groups member of struct device_driver - patch: 10/14 'ASoC: Intel: Select catpt and deprecate haswell' has been moved to the back of the list: enable catpt after machine boards have been prepared for it first - improved readability of several goto labels Changes in v6: https://www.spinics.net/lists/alsa-devel/msg115765.html - reordered and reorganized code for patches 1/13 - 8/13 of v5, so each patches makes use of no member or function which is unavailable to it. Series size increased from 13 to 14 patches: addition of base members e.g.: registers has been split from addition of device.c file which describes acpi device behavior Changes in v5: https://www.spinics.net/lists/alsa-devel/msg115621.html Basically everything below is result of Andy's review. Thank you Andy for taking time into this detailed review - catpt now makes use of common linux/pxa2xx_ssp.h header file, removing redundant SSP register declarations in the process. As stated in the opening, this is dependent upon linux-spi change: spi: pxa2xx: Add SSC2 and SSPSP2 SSP registers - updated Kconfig by removing DMADEVICES and adding COMPILE_TEST as optional depends-on - updated all register macros definitions to be more safe against common arithmetics when specifying macro's parameters - removed CONFIG_PM and CONFIG_PM_SLEEP usage in favor of __maybe_unused - all 'if (ret < 0)' converted to simple 'if (ret)' whenever possible - fixed erroneous check for platform_device_register_data within catpt_register_board() - _SLAVE/_MASTER replaced with more inclusive _CONSUMER/_PROVIDER for enum catpt_ssp_mode - catpt_acpi_probe() is now making use of high-level wrappers for ioremapping and resource assignment, reducing function's code size - due to improved catpt_acpi_probe() behavior, catpt_acpi_remove() needs not to cast dma_free_coherent() any longer - DMA source and destrination maxburst now of value 16, see: https://www.spinics.net/lists/alsa-devel/msg114394.html - simplified catpt_dsp_update_lpclock() as list_for_each_entry() is empty-safe by default - dropped '_SSP_' from all names of all CATPT_SSP_SSXXX_DEFAULT macros - catpt_updatel_pci now makes use of linux/pci.h and uapi/linux/pci.h constants such as: PCI_PM_CTRL_STATE_MASK and PCI_D3hot Changes in v4: https://www.spinics.net/lists/alsa-devel/msg113762.html - fixed compilation with i386 kconfig (conflicting names) - streamlined naming for SHIM and PCI registers to match SSP ones (SHIM_REG -> SHIM) - catpt_component_probe removed and kcontrols again initializzed statically via snd_kcontrol_new array: this is to remove kctl->id.device shenanigans - renamed catpt_set_ctlvol to catpt_set_dspvol - function name wasn't matching its purpose Changes in v3: - fixed IRAM mask usage in lpt_dsp_power_up (dsp.c) - updated dbg message formatting in catpt_restore_fwimage as suggested by Andy - fixed alignment for struct catpt_ssp_device_format - catpt_set_ctlvol now verifies all-equal scenario based on all channels rather than just first two as requested by Amadeo - fixed SPDX for registers.h Changes in v2: https://www.spinics.net/lists/alsa-devel/msg113660.html - fixed SPDX formatting for all header files as well as pcm.c - fixed size provided to memcpy() in fw_build_read() as reported by Mark - renamed struct catpt_pdata to struct catpt_spec (cosmetic) - fixed erroneous path in catpt_load_block: region is properly released - trace.h events for updating registers have been removed and usages replaced by dev_dbg (SRAMPGE/ LPCS) - as requested by Andy, struct resource has replaced struct catpt_mbank and struct catpt_mregion. This change cascaded into: - catpt_mbank_size and catpt_mregion_size replaced by resource_size - catpt_mregion_intersects replaced by resource_overlaps - all catpt_mbank_ and catpt_mregion_ handlers found in loader.c (_request, _reserve, _release, _extract, _split, _join) have been removed - __request_region and __release_region have been enlisted in their place - catpt_mregion_intersecting renamed to catpt_resource_overlapping - catpt_request_region helper has been provided to deal with -size based requests o haven't found direct replacements in resource.c/ ioport.h for both functions - catpt_mbank_create and catpt_mbank_remove renamed to catpt_sram_init and catpt_sram_free respectively - catpt_sram_init now returns void instead of int and has been converted to simple initialized. This change ultimately cascaded into: o both SRAM banks initialization being moved to catpt_dev_init from catpt_acpi_probe (device.c) o catpt_dev::spec is now initialized first, with catpt_dev_init following it soon after o catpt_acpi_probe erroneous path has been simplified as SRAM banks no longer need to be freed - catpt_sram_free now frees all resources via child -> sibling enumeration rather than region_list iteration - catpt_dsp_update_srampge and catpt_dsp_set_srampge now accept new argument: unsigned long mask. Caused by removal of catpt_mbank - mask is taken directly from catpt_dev::spec::d/iram_mask - trace.h events for catpt_mbank and catpt_mregion have been removed Diff against last drop on internal list: https://www.spinics.net/lists/alsa-devel/msg113549.html - replaced spinlock with mutex for mregion allocation and release to address sleeping in atomic context warnings - fixed coredump fw_hash dumping - kcontrol values are now always stored regardless of stream of interest is running or not - kcontrol values are now applied after stream is prepared instead of ignoring what has been set by user initially - catpt_pdata instances have been renamed from hsw_ and bdw_ to lpt_ and wpt_ respectively - reordered Makefile .o(s) (cosmetic) Cezary Rojewski (14): ASoC: Intel: Add catpt base members ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC message handlers ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: PCM operations ASoC: Intel: catpt: Device driver lifecycle ASoC: Intel: catpt: Event tracing ASoC: Intel: catpt: Simple sysfs attributes ASoC: Intel: haswell: Remove haswell-solution specific code ASoC: Intel: broadwell: Remove haswell-solution specific code ASoC: Intel: bdw-5650: Remove haswell-solution specific code ASoC: Intel: bdw-5677: Remove haswell-solution specific code ASoC: Intel: Select catpt and deprecate haswell .../ABI/testing/sysfs-bus-pci-devices-catpt | 16 + sound/soc/intel/Kconfig | 24 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/boards/Kconfig | 8 +- sound/soc/intel/boards/bdw-rt5650.c | 36 - sound/soc/intel/boards/bdw-rt5677.c | 33 - sound/soc/intel/boards/broadwell.c | 33 - sound/soc/intel/boards/haswell.c | 28 +- sound/soc/intel/catpt/Makefile | 6 + sound/soc/intel/catpt/core.h | 187 +++ sound/soc/intel/catpt/device.c | 348 +++++ sound/soc/intel/catpt/dsp.c | 571 ++++++++ sound/soc/intel/catpt/ipc.c | 298 ++++ sound/soc/intel/catpt/loader.c | 671 +++++++++ sound/soc/intel/catpt/messages.c | 313 +++++ sound/soc/intel/catpt/messages.h | 401 ++++++ sound/soc/intel/catpt/pcm.c | 1210 +++++++++++++++++ sound/soc/intel/catpt/registers.h | 178 +++ sound/soc/intel/catpt/sysfs.c | 58 + sound/soc/intel/catpt/trace.h | 83 ++ 20 files changed, 4361 insertions(+), 143 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-catpt create mode 100644 sound/soc/intel/catpt/Makefile create mode 100644 sound/soc/intel/catpt/core.h create mode 100644 sound/soc/intel/catpt/device.c create mode 100644 sound/soc/intel/catpt/dsp.c create mode 100644 sound/soc/intel/catpt/ipc.c create mode 100644 sound/soc/intel/catpt/loader.c create mode 100644 sound/soc/intel/catpt/messages.c create mode 100644 sound/soc/intel/catpt/messages.h create mode 100644 sound/soc/intel/catpt/pcm.c create mode 100644 sound/soc/intel/catpt/registers.h create mode 100644 sound/soc/intel/catpt/sysfs.c create mode 100644 sound/soc/intel/catpt/trace.h -- 2.17.1

4 31

[PATCH] ASoC: hdmi-codec: Use set_jack ops to set jack
by Cheng-Yi Chiang 22 Sep '20

22 Sep '20

Use set_jack ops to set jack so machine drivers do not need to include hdmi-codec.h explicitly. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Separate this patch from the machine driver patch series at https://patchwork.kernel.org/cover/11773209/ in the hope that this can be merged first to ease other machine drivers' usage. include/sound/hdmi-codec.h | 3 --- sound/soc/codecs/hdmi-codec.c | 12 ++++-------- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 ++--- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 5 ++--- .../mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 ++--- sound/soc/rockchip/rockchip_max98090.c | 3 +-- 6 files changed, 11 insertions(+), 22 deletions(-) diff --git a/include/sound/hdmi-codec.h b/include/sound/hdmi-codec.h index 7754631a3102..b55970859a13 100644 --- a/include/sound/hdmi-codec.h +++ b/include/sound/hdmi-codec.h @@ -117,9 +117,6 @@ struct hdmi_codec_pdata { 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 8c6f540533ba..403d4c6a49a8 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -698,13 +698,9 @@ static void plugged_cb(struct device *dev, bool plugged) 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) +static int hdmi_codec_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, + void *data) { struct hdmi_codec_priv *hcp = snd_soc_component_get_drvdata(component); int ret = -EOPNOTSUPP; @@ -720,7 +716,6 @@ int hdmi_codec_set_jack_detect(struct snd_soc_component *component, } return ret; } -EXPORT_SYMBOL_GPL(hdmi_codec_set_jack_detect); static int hdmi_dai_spdif_probe(struct snd_soc_dai *dai) { @@ -806,6 +801,7 @@ static const struct snd_soc_component_driver hdmi_driver = { .use_pmdown_time = 1, .endianness = 1, .non_legacy_dai_naming = 1, + .set_jack = hdmi_codec_set_jack, }; static int hdmi_codec_probe(struct platform_device *pdev) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 347b095d478d..c28ebf891cb0 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -11,7 +11,6 @@ #include <linux/of_gpio.h> #include <sound/soc.h> #include <sound/jack.h> -#include <sound/hdmi-codec.h> #include "../../codecs/rt5645.h" #define MCLK_FOR_CODECS 12288000 @@ -154,8 +153,8 @@ static int mt8173_rt5650_hdmi_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - return hdmi_codec_set_jack_detect(asoc_rtd_to_codec(rtd, 0)->component, - &mt8173_rt5650_hdmi_jack); + return snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, + &mt8173_rt5650_hdmi_jack, NULL); } enum { diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 68fe23b96b14..15e48cde4921 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -10,7 +10,6 @@ #include <linux/module.h> #include <linux/of_device.h> #include <linux/pinctrl/consumer.h> -#include <sound/hdmi-codec.h> #include <sound/jack.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -375,8 +374,8 @@ static int mt8183_da7219_max98357_hdmi_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - return hdmi_codec_set_jack_detect(asoc_rtd_to_codec(rtd, 0)->component, - &priv->hdmi_jack); + return snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, + &priv->hdmi_jack, NULL); } static struct snd_soc_dai_link mt8183_da7219_dai_links[] = { diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 07410d7afaa9..327dfad41e31 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -9,7 +9,6 @@ #include <linux/module.h> #include <linux/of_device.h> #include <linux/pinctrl/consumer.h> -#include <sound/hdmi-codec.h> #include <sound/jack.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -350,8 +349,8 @@ mt8183_mt6358_ts3a227_max98357_hdmi_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - return hdmi_codec_set_jack_detect(asoc_rtd_to_codec(rtd, 0)->component, - &priv->hdmi_jack); + return snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, + &priv->hdmi_jack, NULL); } static struct snd_soc_dai_link mt8183_mt6358_ts3a227_dai_links[] = { diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index 9acfd024aa5d..c8f1a28a92b7 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -12,7 +12,6 @@ #include <linux/gpio.h> #include <linux/of_gpio.h> #include <sound/core.h> -#include <sound/hdmi-codec.h> #include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -238,7 +237,7 @@ static int rk_hdmi_init(struct snd_soc_pcm_runtime *runtime) return ret; } - return hdmi_codec_set_jack_detect(component, &rk_hdmi_jack); + return snd_soc_component_set_jack(component, &rk_hdmi_jack, NULL); } /* max98090 dai_link */ -- 2.28.0.681.g6f77f65b4e-goog

2 1

[PATCH] ALSA: usb-audio: Add mixer support for Pioneer DJ DJM-250MK2
by František Kučera 22 Sep '20

22 Sep '20

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows mapping playback and capture channels to available sources. Configures the card through USB commands. --- sound/usb/mixer_quirks.c | 216 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 216 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199cdbfdc761..8d705d5f60f2 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2602,6 +2602,219 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * Pioneer DJ DJM-250MK2 and maybe other DJM models + * + * For playback, no duplicate mapping should be set. + * There are three mixer stereo channels (CH1, CH2, AUX) + * and three stereo sources (Playback 1-2, Playback 3-4, Playback 5-6). + * Each channel should be mapped just once to one source. + * If mapped multiple times, only one source will play on given channel + * (sources are not mixed together). + * + * For recording, duplicate mapping is OK. We will get the same signal multiple times. + * + * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN. + * + * See also notes in the quirks-table.h file. + */ + +struct snd_pioneer_djm_option { + const u16 wIndex; + const u16 wValue; + const char *name; +}; + +static const struct snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_LEVEL[] = { + { .name = "-5 dB", .wValue = 0x0300, .wIndex = 0x8003 }, + { .name = "-10 dB", .wValue = 0x0200, .wIndex = 0x8003 }, + { .name = "-15 dB", .wValue = 0x0100, .wIndex = 0x8003 }, + { .name = "-19 dB", .wValue = 0x0000, .wIndex = 0x8003 } +}; + +static const struct snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH12[] = { + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0109, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x010d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x010a, .wIndex = 0x8002 } +}; + +static const struct snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH34[] = { + { .name = "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0209, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x020d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x020a, .wIndex = 0x8002 } +}; + +static const struct snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH56[] = { + { .name = "REC OUT", .wValue = 0x030a, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0311, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0312, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0309, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x030d, .wIndex = 0x8002 } +}; + +static const struct snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_12[] = { + { .name = "CH1", .wValue = 0x0100, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0101, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0104, .wIndex = 0x8016 } +}; + +static const struct snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_34[] = { + { .name = "CH1", .wValue = 0x0200, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0201, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0204, .wIndex = 0x8016 } +}; + +static const struct snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_56[] = { + { .name = "CH1", .wValue = 0x0300, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0301, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0304, .wIndex = 0x8016 } +}; + +struct snd_pioneer_djm_option_group { + const char *name; + const struct snd_pioneer_djm_option *options; + const size_t count; + const u16 default_value; +} snd_pioneer_djm_option_group; + +#define SND_PIONEER_DJM_OPTION_GROUP_ITEM(_name, suffix, _default_value) { \ + .name = _name, \ + .options = SND_PIONEER_DJM_OPTIONS_##suffix, \ + .count = ARRAY_SIZE(SND_PIONEER_DJM_OPTIONS_##suffix), \ + .default_value = _default_value } + +static const struct snd_pioneer_djm_option_group SND_PIONEER_DJM_OPTION_GROUPS[] = { + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Master Capture Level Capture Switch", CAPTURE_LEVEL, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 1-2 Capture Switch", CAPTURE_CH12, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 3-4 Capture Switch", CAPTURE_CH34, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 5-6 Capture Switch", CAPTURE_CH56, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 1-2 Playback Switch", PLAYBACK_12, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 3-4 Playback Switch", PLAYBACK_34, 1), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 5-6 Playback Switch", PLAYBACK_56, 2) +}; + +// layout of the kcontrol->private_value: +#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff +#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000 +#define SND_PIONEER_DJM_GROUP_SHIFT 16 + +static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *info) +{ + u16 group_index = kctl->private_value >> SND_PIONEER_DJM_GROUP_SHIFT; + size_t count; + const char *name; + const struct snd_pioneer_djm_option_group *group; + + if (group_index < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS)) { + group = &SND_PIONEER_DJM_OPTION_GROUPS[group_index]; + count = group->count; + if (info->value.enumerated.item >= count) + info->value.enumerated.item = count - 1; + name = group->options[info->value.enumerated.item].name; + strlcpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name)); + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = count; + return 0; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value) +{ + int err; + + if (group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS) + && value < SND_PIONEER_DJM_OPTION_GROUPS[group].count) { + err = snd_usb_lock_shutdown(mixer->chip); + if (err) + return err; + + err = snd_usb_ctl_msg( + mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), + USB_REQ_SET_FEATURE, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wValue, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wIndex, + NULL, 0); + + snd_usb_unlock_shutdown(mixer->chip); + return err; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + elem->value.enumerated.item[0] = kctl->private_value & SND_PIONEER_DJM_VALUE_MASK; + return 0; +} + +static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + unsigned long private_value = kctl->private_value; + + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = elem->value.enumerated.item[0]; + + kctl->private_value = group << SND_PIONEER_DJM_GROUP_SHIFT | value; + + return snd_pioneer_djm_controls_update(mixer, group, value); +} + +static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list) +{ + unsigned long private_value = list->kctl->private_value; + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = (private_value & SND_PIONEER_DJM_VALUE_MASK); + + return snd_pioneer_djm_controls_update(list->mixer, group, value); +} + +static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + const struct snd_pioneer_djm_option_group *group; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_pioneer_djm_controls_info, + .get = snd_pioneer_djm_controls_get, + .put = snd_pioneer_djm_controls_put + }; + + for (i = 0; i < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS); i++) { + group = &SND_PIONEER_DJM_OPTION_GROUPS[i]; + knew.name = group->name; + knew.private_value = i << SND_PIONEER_DJM_GROUP_SHIFT | group->default_value; + err = snd_pioneer_djm_controls_update(mixer, i, group->default_value); + if (err) + return err; + err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume, + &knew, NULL); + if (err) + return err; + } + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2706,6 +2919,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ + err = snd_pioneer_djm_controls_create(mixer); + break; } return err; -- 2.20.1

2 1

[PATCH 1/9] ASoC: tas2770: Fix calling reset in probe
by Dan Murphy 22 Sep '20

22 Sep '20

tas2770_reset is called during i2c probe. The reset calls the snd_soc_component_write which depends on the tas2770->component being available. The component pointer is not set until codec_probe so move the reset to the codec_probe after the pointer is set. Fixes: 1a476abc723e6 ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2770.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index c09851834395..03d7ad1885b8 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -575,6 +575,8 @@ static int tas2770_codec_probe(struct snd_soc_component *component) tas2770->component = component; + tas2770_reset(tas2770); + return 0; } @@ -771,8 +773,6 @@ static int tas2770_i2c_probe(struct i2c_client *client, 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"); -- 2.28.0

2 13

Bad vumeter data in aplay/arecord
by GitHub issues - opened 22 Sep '20

22 Sep '20

alsa-project/alsa-utils issue #57 was opened from SuitableUsername: In aplay.c the wrong count is passed to compute_max_peak(). Instead of "r * hwparams.channels" it should be "r * bits_per_frame / 8". Also, the vumeter should really show an overall peak level percentage because the transitory 1/8 second one has very little value. Issue URL : https://github.com/alsa-project/alsa-utils/issues/57 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: ctl: Workaround for lockdep warning wrt card->ctl_files_rwlock
by Takashi Iwai 22 Sep '20

22 Sep '20

The recent change in lockdep for read lock caused the deadlock warnings in ALSA control code which uses the read_lock() for notification and else while write_lock_irqsave() is used for adding and removing the list entry. Although a deadlock would practically never hit in a real usage (the addition and the deletion can't happen with the notification), it's better to fix the read_lock() usage in a semantically correct way. This patch replaces the read_lock() calls with read_lock_irqsave() version for avoiding a reported deadlock. The notification code path takes the irq disablement in anyway, and other code paths are very short execution, hence there shouldn't be any big performance hit by this change. Fixes: e918188611f0 ("locking: More accurate annotations for read_lock()") Reported-by: syzbot+561a74f84100162990b2(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index e014598142df..421ddc76f264 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -150,14 +150,14 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, return; if (card->shutdown) return; - read_lock(&card->ctl_files_rwlock); + read_lock_irqsave(&card->ctl_files_rwlock, flags); #if IS_ENABLED(CONFIG_SND_MIXER_OSS) card->mixer_oss_change_count++; #endif list_for_each_entry(ctl, &card->ctl_files, list) { if (!ctl->subscribed) continue; - spin_lock_irqsave(&ctl->read_lock, flags); + spin_lock(&ctl->read_lock); list_for_each_entry(ev, &ctl->events, list) { if (ev->id.numid == id->numid) { ev->mask |= mask; @@ -174,10 +174,10 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, } _found: wake_up(&ctl->change_sleep); - spin_unlock_irqrestore(&ctl->read_lock, flags); + spin_unlock(&ctl->read_lock); kill_fasync(&ctl->fasync, SIGIO, POLL_IN); } - read_unlock(&card->ctl_files_rwlock); + read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } EXPORT_SYMBOL(snd_ctl_notify); @@ -1951,8 +1951,9 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type) { struct snd_ctl_file *kctl; int subdevice = -1; + unsigned long flags; - read_lock(&card->ctl_files_rwlock); + read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(kctl, &card->ctl_files, list) { if (kctl->pid == task_pid(current)) { subdevice = kctl->preferred_subdevice[type]; @@ -1960,7 +1961,7 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type) break; } } - read_unlock(&card->ctl_files_rwlock); + read_unlock_irqrestore(&card->ctl_files_rwlock, flags); return subdevice; } EXPORT_SYMBOL_GPL(snd_ctl_get_preferred_subdevice); @@ -2009,13 +2010,14 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) { struct snd_card *card = device->device_data; struct snd_ctl_file *ctl; + unsigned long flags; - read_lock(&card->ctl_files_rwlock); + read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); } - read_unlock(&card->ctl_files_rwlock); + read_unlock_irqrestore(&card->ctl_files_rwlock, flags); return snd_unregister_device(&card->ctl_dev); } -- 2.16.4

1 0

[PATCH 00/20] Add support for SATA/PCIe/USB2[3]/VIN/CSI on R8A774E1
by Lad Prabhakar 22 Sep '20

22 Sep '20

Hi All, This patch series adds support for the following peripherals on RZ/G2H SoC * PCIe * SATA * USB2 * USB3 * Audio * VIN * CSI Cheers, Prabhakar Lad Prabhakar (20): dt-bindings: pci: rcar-pci: Add device tree support for r8a774e1 arm64: dts: renesas: r8a774e1: Add PCIe device nodes dt-bindings: ata: renesas,rcar-sata: Add r8a774e1 support arm64: dts: renesas: r8a774e1: Add SATA controller node dt-bindings: phy: renesas,usb2-phy: Add r8a774e1 support arm64: dts: renesas: r8a774e1: Add USB2.0 phy and host (EHCI/OHCI) device nodes dt-bindings: usb: renesas,usb3-peri: Document r8a774e1 support dt-bindings: usb: usb-xhci: Document r8a774e1 support dt-bindings: phy: renesas,usb3-phy: Add r8a774e1 support arm64: dts: renesas: r8a774e1: Add USB3.0 device nodes dt-bindings: usb: renesas,usbhs: Add r8a774e1 support dt-bindings: dma: renesas,usb-dmac: Add binding for r8a774e1 arm64: dts: renesas: r8a774e1: Add USB-DMAC and HSUSB device nodes dt-bindings: sound: renesas,rsnd: Document r8a774e1 bindings arm64: dts: renesas: r8a774e1: Add audio support dt-bindings: media: renesas,csi2: Add R8A774E1 support dt-bindings: media: renesas,vin: Add R8A774E1 support media: rcar-csi2: Enable support for R8A774E1 media: rcar-vin: Enable support for R8A774E1 arm64: dts: renesas: r8a774e1: Add VIN and CSI-2 nodes .../bindings/ata/renesas,rcar-sata.yaml | 1 + .../bindings/dma/renesas,usb-dmac.yaml | 1 + .../bindings/media/renesas,csi2.yaml | 1 + .../bindings/media/renesas,vin.yaml | 1 + .../devicetree/bindings/pci/rcar-pci.txt | 1 + .../bindings/phy/renesas,usb2-phy.yaml | 1 + .../bindings/phy/renesas,usb3-phy.yaml | 1 + .../bindings/sound/renesas,rsnd.txt | 1 + .../bindings/usb/renesas,usb3-peri.yaml | 1 + .../bindings/usb/renesas,usbhs.yaml | 1 + .../devicetree/bindings/usb/usb-xhci.txt | 1 + arch/arm64/boot/dts/renesas/r8a774e1.dtsi | 989 +++++++++++++++++- drivers/media/platform/rcar-vin/rcar-core.c | 40 + drivers/media/platform/rcar-vin/rcar-csi2.c | 4 + 14 files changed, 1022 insertions(+), 22 deletions(-) -- 2.17.1

11 77

[PATCH] ALSA: seq: oss: Avoid mutex lock for a long-time ioctl
by Takashi Iwai 22 Sep '20

22 Sep '20

Recently we applied a fix to cover the whole OSS sequencer ioctls with the mutex for dealing with the possible races. This works fine in general, but in theory, this may lead to unexpectedly long stall if an ioctl like SNDCTL_SEQ_SYNC is issued and an event with the far future timestamp was queued. For fixing such a potential stall, this patch changes the mutex lock applied conditionally excluding such an ioctl command. Also, change the mutex_lock() with the interruptible version for user to allow escaping from the big-hammer mutex. Fixes: 80982c7e834e ("ALSA: seq: oss: Serialize ioctls") Suggested-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c index c8b9c0b315d8..250a92b18726 100644 --- a/sound/core/seq/oss/seq_oss.c +++ b/sound/core/seq/oss/seq_oss.c @@ -174,9 +174,12 @@ odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg) if (snd_BUG_ON(!dp)) return -ENXIO; - mutex_lock(&register_mutex); + if (cmd != SNDCTL_SEQ_SYNC && + mutex_lock_interruptible(&register_mutex)) + return -ERESTARTSYS; rc = snd_seq_oss_ioctl(dp, cmd, arg); - mutex_unlock(&register_mutex); + if (cmd != SNDCTL_SEQ_SYNC) + mutex_unlock(&register_mutex); return rc; } -- 2.16.4

1 0

[PATCH] ASoC: hdac: make SOF HDA codec driver probe deterministic
by Kai Vehmanen 22 Sep '20

22 Sep '20

To provide backward compatibility to older systems, the SOF HDA driver allows user to specify which HDMI codec driver to use at runtime via kernel parameter. This mechanism has a subtle flaw in that it assumes the codec drivers not to be loaded when the SOF PCI driver is loaded. The problem is rooted in use of the hdev->type field. snd_hdac_ext_bus_device_init() initializes this field to HDA_DEV_ASOC. This signals the HDA core that ASoC drivers should be considered in driver matching (hda_bus_match()). The SOF and SST drivers continue by overriding this field to HDA_DEV_LEGACY and proceeding to load driver modules with request_module(). Correct drivers will get loaded and attached. If however the codec drivers are already loaded when snd_hdac_ext_bus_device_init() is called, the matching will not work as expected as device type is still set to HDA_DEV_ASOC. Specifically if hdac-hdmi is attached when machine driver is configured to use hdac-hda, this leads to out-of-bounds memory access in hda_dsp_hdmi_build_controls(). Fix the issue by adding codec type as a parameter to snd_hdac_ext_bus_device_init() and ensuring type is set correctly from the start. BugLink: https://github.com/thesofproject/linux/issues/2311 Fixes: 139c7febad1a ("ASoC: SOF: Intel: add support for snd-hda-codec-hdmi") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/hdaudio_ext.h | 2 +- sound/hda/ext/hdac_ext_bus.c | 5 +++-- sound/soc/intel/skylake/skl.c | 4 ++-- sound/soc/sof/intel/hda-codec.c | 17 ++++++++--------- 4 files changed, 14 insertions(+), 14 deletions(-) diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index ef88b20c7b0a..7abf74c1c474 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -10,7 +10,7 @@ int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, void snd_hdac_ext_bus_exit(struct hdac_bus *bus); int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, - struct hdac_device *hdev); + struct hdac_device *hdev, int type); void snd_hdac_ext_bus_device_exit(struct hdac_device *hdev); void snd_hdac_ext_bus_device_remove(struct hdac_bus *bus); diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index d0a604c939df..765c40a6ccba 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -70,11 +70,12 @@ static void default_release(struct device *dev) * @bus: hdac bus to attach to * @addr: codec address * @hdev: hdac device to init + * @type: codec type (HDAC_DEV_*) to use for this device * * Returns zero for success or a negative error code. */ int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, - struct hdac_device *hdev) + struct hdac_device *hdev, int type) { char name[15]; int ret; @@ -88,7 +89,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_bus *bus, int addr, dev_err(bus->dev, "device init failed for hdac device\n"); return ret; } - hdev->type = HDA_DEV_ASOC; + hdev->type = type; hdev->dev.release = default_release; ret = snd_hdac_device_register(hdev); diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index ea00cf61d1a8..8b993722f74e 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -721,7 +721,7 @@ static int probe_codec(struct hdac_bus *bus, int addr) hda_codec->codec.bus = skl_to_hbus(skl); hdev = &hda_codec->codec.core; - err = snd_hdac_ext_bus_device_init(bus, addr, hdev); + err = snd_hdac_ext_bus_device_init(bus, addr, hdev, HDA_DEV_ASOC); if (err < 0) return err; @@ -736,7 +736,7 @@ static int probe_codec(struct hdac_bus *bus, int addr) if (!hdev) return -ENOMEM; - return snd_hdac_ext_bus_device_init(bus, addr, hdev); + return snd_hdac_ext_bus_device_init(bus, addr, hdev, HDA_DEV_ASOC); #endif /* CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC */ } diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index f6ba3b593e1f..6875fa570c2c 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -117,6 +117,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) struct hdac_hda_priv *hda_priv; struct hda_codec *codec; + int type = HDA_DEV_LEGACY; #endif struct hda_bus *hbus = sof_to_hbus(sdev); struct hdac_device *hdev; @@ -143,7 +144,11 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, hdev = &hda_priv->codec.core; codec = &hda_priv->codec; - ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev); + /* only probe ASoC codec drivers for HDAC-HDMI */ + if (!hda_codec_use_common_hdmi && (resp & 0xFFFF0000) == IDISP_VID_INTEL) + type = HDA_DEV_ASOC; + + ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev, type); if (ret < 0) return ret; @@ -161,13 +166,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, else codec->probe_id = 0; - /* - * if common HDMI codec driver is not used, codec load - * is skipped here and hdac_hdmi is used instead - */ - if (hda_codec_use_common_hdmi || - (resp & 0xFFFF0000) != IDISP_VID_INTEL) { - hdev->type = HDA_DEV_LEGACY; + if (type == HDA_DEV_LEGACY) { ret = hda_codec_load_module(codec); /* * handle ret==0 (no driver bound) as an error, but pass @@ -188,7 +187,7 @@ static int hda_codec_probe(struct snd_sof_dev *sdev, int address, if (!hdev) return -ENOMEM; - ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev); + ret = snd_hdac_ext_bus_device_init(&hbus->core, address, hdev, HDA_DEV_ASOC); return ret; #endif -- 2.27.0

2 1

[PATCH] dt-bindings: tas2770: Fix I2C addresses for the TAS2770
by Dan Murphy 22 Sep '20

22 Sep '20

The I2C addresses listed in the yaml are not correct. The addresses can range from 0x41 through 0x48 based on register configurations. Fix the example and the description. Fixes: 4b7151dadfd4 ("dt-bindings: ASoC: Add tas2770 smart PA dt bindings") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tas2770.yaml | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml index 33a90f829c80..bb26d081c9fa 100644 --- a/Documentation/devicetree/bindings/sound/tas2770.yaml +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -24,7 +24,7 @@ properties: reg: maxItems: 1 description: | - I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + I2C address of the device can be between 0x41 to 0x48. reset-gpio: description: GPIO used to reset the device. @@ -62,9 +62,9 @@ examples: i2c0 { #address-cells = <1>; #size-cells = <0>; - codec: codec@4c { + codec: codec@41 { compatible = "ti,tas2770"; - reg = <0x4c>; + reg = <0x41>; #sound-dai-cells = <1>; interrupt-parent = <&gpio1>; interrupts = <14>; -- 2.28.0

2 2

[PATCH 1/2] dt-bindings: tas2562: Add the TAS2110 amplifier
by Dan Murphy 22 Sep '20

22 Sep '20

Add the TAS2110 amplifier compatible. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- This patchset is based on top of patchset https://lore.kernel.org/patchwork/project/lkml/list/?series=463709 Documentation/devicetree/bindings/sound/tas2562.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index d3ceda356aaa..27f7132ba2ef 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -20,6 +20,7 @@ description: | https://www.ti.com/lit/gpn/tas2562 https://www.ti.com/lit/gpn/tas2563 https://www.ti.com/lit/gpn/tas2564 + https://www.ti.com/lit/gpn/tas2110 properties: compatible: @@ -27,6 +28,7 @@ properties: - ti,tas2562 - ti,tas2563 - ti,tas2564 + - ti,tas2110 reg: maxItems: 1 -- 2.28.0

2 2

[PATCH 0/3] tlv320aic3xx4 updates
by Miquel Raynal 21 Sep '20

21 Sep '20

Hello, While doing a kernel update on a sama5-based board I figured out the sound system was not working anymore, the debug session led me to the following commits. As I am not an audio expert at all, I am fully open to comments and suggestions. Thanks, Miquèl Miquel Raynal (3): ASoC: tlv320aic32x4: Ensure a minimum delay before clock stabilization ASoC: tlv320aic32x4: Fix bdiv clock rate derivation ASoC: tlv320aic32x4: Enable fast charge sound/soc/codecs/tlv320aic32x4-clk.c | 9 ++++++++- sound/soc/codecs/tlv320aic32x4.c | 17 ++++++++++++++--- sound/soc/codecs/tlv320aic32x4.h | 7 +++++++ 3 files changed, 29 insertions(+), 4 deletions(-) -- 2.20.1

3 11

[PATCH v6 00/14] ASoC: Intel: Catpt - Lynx and Wildcat point
by Cezary Rojewski 21 Sep '20

21 Sep '20

Implement support for Lynxpoint and Wildcat Point AudioDSP. Catpt solution deprecates existing sound/soc/intel/haswell which is removed in the following series. This cover-letter is followed by 'Developer's deep dive' message schedding light on catpt's key concepts and areas addressed. Due to high range of errors and desynchronization from recommendations set by Windows solution, re-write came as a lower-cost solution compared to refactoring /haswell/ with several series of patches. Series is dependent on linux-spi change: spi: pxa2xx: Add SSC2 and SSPSP2 SSP registers https://www.spinics.net/lists/linux-spi/msg23885.html which has been already merged and is now part of linux-spi tree. Bulk of series content is device driver core code - everything up to patch 7/14 - with fs entries and trace macros introduced right after. While each core patch is shaped in such a way that no unavailable members are ever called, until patch 10/14 is applied, no code compilation can occur as no Makefile is present. Once said patch is added, Makefile and Kconfig are implemented and driver module compiles as expected. Special thanks go to Marcin Barlik and Piotr Papierkowski for sharing their LPT/WPT AudioDSP architecture expertise as well as helping backtrack its historical background. My thanks go to Amadeusz Slawinski for reviews and improvements proposed on and off the internal list. Most of internal diff below is his contribution. Krzysztof Hejmowski helped me setup my own Xtensa environment and recompile LPT/WPT FW binary sources what sped up the development greatly. This would not have been possible without help from these champions, especially considering how quickly the catpt was written: 2 weeks features, 3 weeks optimizations. Thank you. Userspace-exposed members are compatible with what is exposed by deprecated solution as well as FW binary being re-used thus no harm is done. The only visible differences are: the newly added 'Loopback Mute' kcontrol and volume support extending to quad from stereo. On top of fixing erros and design flows, catpt also adds module reload, dynamic SRAM memory allocation during PCM runtime and exposes missing userspace API: 'Loopback Mute' kcontrol, quad volume controls and sysfs fw-version entries. Event tracing is provided to easy solution debugging. Following are not included in this update and are scheduled as later addition: - fw logging - module (library) support Note: LPT power up/down sequences might get aligned with WPT once enough testing is done as capabilities are shared for both DSPs. Note #2: Both LPT and WPT power up/down sequences may get optimized in future updates as thanks to help from the Windows team, most of nuances behind why/what/when in regard to hw registers have been backtracked and reviewed again. Link to developer's deep dive message: https://www.spinics.net/lists/alsa-devel/msg113563.html Changes in v6: - reordered and reorganized code for patches 1/13 - 8/13 of v5, so each patches makes use of no member or function which is unavailable to it. Series size increased from 13 to 14 patches: addition of base members e.g.: registers has been split from addition of device.c file which describes acpi device behavior Changes in v5: https://www.spinics.net/lists/alsa-devel/msg115621.html Basically everything below is result of Andy's review. Thank you Andy for taking time into this detailed review - catpt now makes use of common linux/pxa2xx_ssp.h header file, removing redundant SSP register declarations in the process. As stated in the opening, this is dependent upon linux-spi change: spi: pxa2xx: Add SSC2 and SSPSP2 SSP registers - updated Kconfig by removing DMADEVICES and adding COMPILE_TEST as optional depends-on - updated all register macros definitions to be more safe against common arithmetics when specifying macro's parameters - removed CONFIG_PM and CONFIG_PM_SLEEP usage in favor of __maybe_unused - all 'if (ret < 0)' converted to simple 'if (ret)' whenever possible - fixed erroneous check for platform_device_register_data within catpt_register_board() - _SLAVE/_MASTER replaced with more inclusive _CONSUMER/_PROVIDER for enum catpt_ssp_mode - catpt_acpi_probe() is now making use of high-level wrappers for ioremapping and resource assignment, reducing function's code size - due to improved catpt_acpi_probe() behavior, catpt_acpi_remove() needs not to cast dma_free_coherent() any longer - DMA source and destrination maxburst now of value 16, see: https://www.spinics.net/lists/alsa-devel/msg114394.html - simplified catpt_dsp_update_lpclock() as list_for_each_entry() is empty-safe by default - dropped '_SSP_' from all names of all CATPT_SSP_SSXXX_DEFAULT macros - catpt_updatel_pci now makes use of linux/pci.h and uapi/linux/pci.h constants such as: PCI_PM_CTRL_STATE_MASK and PCI_D3hot Changes in v4: https://www.spinics.net/lists/alsa-devel/msg113762.html - fixed compilation with i386 kconfig (conflicting names) - streamlined naming for SHIM and PCI registers to match SSP ones (SHIM_REG -> SHIM) - catpt_component_probe removed and kcontrols again initializzed statically via snd_kcontrol_new array: this is to remove kctl->id.device shenanigans - renamed catpt_set_ctlvol to catpt_set_dspvol - function name wasn't matching its purpose Changes in v3: - fixed IRAM mask usage in lpt_dsp_power_up (dsp.c) - updated dbg message formatting in catpt_restore_fwimage as suggested by Andy - fixed alignment for struct catpt_ssp_device_format - catpt_set_ctlvol now verifies all-equal scenario based on all channels rather than just first two as requested by Amadeo - fixed SPDX for registers.h Changes in v2: https://www.spinics.net/lists/alsa-devel/msg113660.html - fixed SPDX formatting for all header files as well as pcm.c - fixed size provided to memcpy() in fw_build_read() as reported by Mark - renamed struct catpt_pdata to struct catpt_spec (cosmetic) - fixed erroneous path in catpt_load_block: region is properly released - trace.h events for updating registers have been removed and usages replaced by dev_dbg (SRAMPGE/ LPCS) - as requested by Andy, struct resource has replaced struct catpt_mbank and struct catpt_mregion. This change cascaded into: - catpt_mbank_size and catpt_mregion_size replaced by resource_size - catpt_mregion_intersects replaced by resource_overlaps - all catpt_mbank_ and catpt_mregion_ handlers found in loader.c (_request, _reserve, _release, _extract, _split, _join) have been removed - __request_region and __release_region have been enlisted in their place - catpt_mregion_intersecting renamed to catpt_resource_overlapping - catpt_request_region helper has been provided to deal with -size based requests o haven't found direct replacements in resource.c/ ioport.h for both functions - catpt_mbank_create and catpt_mbank_remove renamed to catpt_sram_init and catpt_sram_free respectively - catpt_sram_init now returns void instead of int and has been converted to simple initialized. This change ultimately cascaded into: o both SRAM banks initialization being moved to catpt_dev_init from catpt_acpi_probe (device.c) o catpt_dev::spec is now initialized first, with catpt_dev_init following it soon after o catpt_acpi_probe erroneous path has been simplified as SRAM banks no longer need to be freed - catpt_sram_free now frees all resources via child -> sibling enumeration rather than region_list iteration - catpt_dsp_update_srampge and catpt_dsp_set_srampge now accept new argument: unsigned long mask. Caused by removal of catpt_mbank - mask is taken directly from catpt_dev::spec::d/iram_mask - trace.h events for catpt_mbank and catpt_mregion have been removed Diff against last drop on internal list: https://www.spinics.net/lists/alsa-devel/msg113549.html - replaced spinlock with mutex for mregion allocation and release to address sleeping in atomic context warnings - fixed coredump fw_hash dumping - kcontrol values are now always stored regardless of stream of interest is running or not - kcontrol values are now applied after stream is prepared instead of ignoring what has been set by user initially - catpt_pdata instances have been renamed from hsw_ and bdw_ to lpt_ and wpt_ respectively - reordered Makefile .o(s) (cosmetic) Cezary Rojewski (14): ASoC: Intel: Add catpt base members ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC message handlers ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: PCM operations ASoC: Intel: catpt: Device driver lifecycle ASoC: Intel: catpt: Event tracing ASoC: Intel: catpt: Simple sysfs attributes ASoC: Intel: Select catpt and deprecate haswell ASoC: Intel: haswell: Remove haswell-solution specific code ASoC: Intel: broadwell: Remove haswell-solution specific code ASoC: Intel: bdw-5650: Remove haswell-solution specific code ASoC: Intel: bdw-5677: Remove haswell-solution specific code sound/soc/intel/Kconfig | 24 +- sound/soc/intel/Makefile | 2 +- sound/soc/intel/boards/Kconfig | 8 +- sound/soc/intel/boards/bdw-rt5650.c | 36 - sound/soc/intel/boards/bdw-rt5677.c | 33 - sound/soc/intel/boards/broadwell.c | 33 - sound/soc/intel/boards/haswell.c | 28 +- sound/soc/intel/catpt/Makefile | 6 + sound/soc/intel/catpt/core.h | 189 +++++ sound/soc/intel/catpt/device.c | 354 ++++++++ sound/soc/intel/catpt/dsp.c | 572 +++++++++++++ sound/soc/intel/catpt/fs.c | 79 ++ sound/soc/intel/catpt/ipc.c | 298 +++++++ sound/soc/intel/catpt/loader.c | 671 +++++++++++++++ sound/soc/intel/catpt/messages.c | 313 +++++++ sound/soc/intel/catpt/messages.h | 401 +++++++++ sound/soc/intel/catpt/pcm.c | 1212 +++++++++++++++++++++++++++ sound/soc/intel/catpt/registers.h | 178 ++++ sound/soc/intel/catpt/trace.h | 83 ++ 19 files changed, 4377 insertions(+), 143 deletions(-) create mode 100644 sound/soc/intel/catpt/Makefile create mode 100644 sound/soc/intel/catpt/core.h create mode 100644 sound/soc/intel/catpt/device.c create mode 100644 sound/soc/intel/catpt/dsp.c create mode 100644 sound/soc/intel/catpt/fs.c create mode 100644 sound/soc/intel/catpt/ipc.c create mode 100644 sound/soc/intel/catpt/loader.c create mode 100644 sound/soc/intel/catpt/messages.c create mode 100644 sound/soc/intel/catpt/messages.h create mode 100644 sound/soc/intel/catpt/pcm.c create mode 100644 sound/soc/intel/catpt/registers.h create mode 100644 sound/soc/intel/catpt/trace.h -- 2.17.1

5 23

[PATCH v2 0/2] DAI driver for new XCVR IP
by Viorel Suman (OSS) 21 Sep '20

21 Sep '20

From: Viorel Suman <viorel.suman(a)nxp.com> DAI driver for new XCVR IP found in i.MX8MP. Viorel Suman (2): ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver ASoC: dt-bindings: fsl_xcvr: Add document for XCVR Changes since v1: - improved 6- and 12-ch layout comment - used regmap polling function, improved clocks handling in runtime_resume - added FW size check in FW load function, improved IRQ handler, removed dummy IRQ handlers - fixed yaml file .../devicetree/bindings/sound/fsl,xcvr.yaml | 103 ++ sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_xcvr.c | 1343 ++++++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 266 ++++ 5 files changed, 1724 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,xcvr.yaml create mode 100644 sound/soc/fsl/fsl_xcvr.c create mode 100644 sound/soc/fsl/fsl_xcvr.h -- 2.7.4

1 2

[PATCH] ALSA: usb-audio: Add mixer support for Pioneer DJ DJM-250MK2
by František Kučera 21 Sep '20

21 Sep '20

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows mapping playback and capture channels to available sources. Configures the card through USB commands. --- sound/usb/mixer_quirks.c | 216 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 216 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199cdbfdc761..b7816a71c42d 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2602,6 +2602,219 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * Pioneer DJ DJM-250MK2 and maybe other DJM models + * + * For playback, no duplicate mapping should be set. + * There are three mixer stereo channels (CH1, CH2, AUX) + * and three stereo sources (Playback 1-2, Playback 3-4, Playback 5-6). + * Each channel should be mapped just once to one source. + * If mapped multiple times, only one source will play on given channel + * (sources are not mixed together). + * + * For recording, duplicate mapping is OK. We will get the same signal multiple times. + * + * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN. + * + * See also notes in the quirks-table.h file. + */ + +typedef struct { + const u16 wIndex; + const u16 wValue; + const char *name; +} snd_pioneer_djm_option; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_LEVEL[] = { + { .name = "-5 dB", .wValue = 0x0300, .wIndex = 0x8003 }, + { .name = "-10 dB", .wValue = 0x0200, .wIndex = 0x8003 }, + { .name = "-15 dB", .wValue = 0x0100, .wIndex = 0x8003 }, + { .name = "-19 dB", .wValue = 0x0000, .wIndex = 0x8003 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH12[] = { + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0109, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x010d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x010a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH34[] = { + { .name = "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0209, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x020d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x020a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH56[] = { + { .name = "REC OUT", .wValue = 0x030a, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0311, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0312, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0309, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x030d, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_12[] = { + { .name = "CH1", .wValue = 0x0100, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0101, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0104, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_34[] = { + { .name = "CH1", .wValue = 0x0200, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0201, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0204, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_56[] = { + { .name = "CH1", .wValue = 0x0300, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0301, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0304, .wIndex = 0x8016 } +}; + +typedef struct { + const char *name; + const snd_pioneer_djm_option *options; + const size_t count; + const u16 default_value; +} snd_pioneer_djm_option_group; + +#define SND_PIONEER_DJM_OPTION_GROUP_ITEM(_name, suffix, _default_value) { \ + .name = _name, \ + .options = SND_PIONEER_DJM_OPTIONS_##suffix, \ + .count = ARRAY_SIZE(SND_PIONEER_DJM_OPTIONS_##suffix), \ + .default_value = _default_value } + +static const snd_pioneer_djm_option_group SND_PIONEER_DJM_OPTION_GROUPS[] = { + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Master Capture Level Capture Switch", CAPTURE_LEVEL, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 1-2 Capture Switch", CAPTURE_CH12, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 3-4 Capture Switch", CAPTURE_CH34, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 5-6 Capture Switch", CAPTURE_CH56, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 1-2 Playback Switch", PLAYBACK_12, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 3-4 Playback Switch", PLAYBACK_34, 1), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 5-6 Playback Switch", PLAYBACK_56, 2) +}; + +// layout of the kcontrol->private_value: +#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff +#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000 +#define SND_PIONEER_DJM_GROUP_SHIFT 16 + +static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *info) +{ + u16 group_index = kctl->private_value >> SND_PIONEER_DJM_GROUP_SHIFT; + size_t count; + const char *name; + const snd_pioneer_djm_option_group *group; + + if (group_index < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS)) { + group = &SND_PIONEER_DJM_OPTION_GROUPS[group_index]; + count = group->count; + if (info->value.enumerated.item >= count) + info->value.enumerated.item = count - 1; + name = group->options[info->value.enumerated.item].name; + strlcpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name)); + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = count; + return 0; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value) +{ + int err; + + if (group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS) + && value < SND_PIONEER_DJM_OPTION_GROUPS[group].count) { + err = snd_usb_lock_shutdown(mixer->chip); + if (err) + return err; + + err = snd_usb_ctl_msg( + mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), + USB_REQ_SET_FEATURE, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wValue, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wIndex, + NULL, 0); + + snd_usb_unlock_shutdown(mixer->chip); + return err; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + elem->value.enumerated.item[0] = kctl->private_value & SND_PIONEER_DJM_VALUE_MASK; + return 0; +} + +static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + unsigned long private_value = kctl->private_value; + + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = elem->value.enumerated.item[0]; + + kctl->private_value = group << SND_PIONEER_DJM_GROUP_SHIFT | value; + + return snd_pioneer_djm_controls_update(mixer, group, value); +} + +static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list) +{ + unsigned long private_value = list->kctl->private_value; + u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = (private_value & SND_PIONEER_DJM_VALUE_MASK); + + return snd_pioneer_djm_controls_update(list->mixer, group, value); +} + +static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) +{ + int err, i; + const snd_pioneer_djm_option_group *group; + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_pioneer_djm_controls_info, + .get = snd_pioneer_djm_controls_get, + .put = snd_pioneer_djm_controls_put + }; + + for (i = 0; i < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS); i++) { + group = &SND_PIONEER_DJM_OPTION_GROUPS[i]; + knew.name = group->name; + knew.private_value = i << SND_PIONEER_DJM_GROUP_SHIFT | group->default_value; + err = snd_pioneer_djm_controls_update(mixer, i, group->default_value); + if (err) + return err; + err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume, + &knew, NULL); + if (err) + return err; + } + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2706,6 +2919,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ + err = snd_pioneer_djm_controls_create(mixer); + break; } return err; -- 2.20.1

1 0

[PATCH] ASoC: SOF: pm: Fix prepare callback behavior for OF usecase
by Kai Vehmanen 21 Sep '20

21 Sep '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> On i.MX platforms PM is not managed via ACPI although CONFIG_ACPI can be set. So, in order to correctly set the system target state we introduce a flag for platforms that require to use acpi target states. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/sof.h | 2 ++ sound/soc/sof/pm.c | 12 +++++++----- sound/soc/sof/sof-pci-dev.c | 9 +++++++++ 3 files changed, 18 insertions(+), 5 deletions(-) diff --git a/include/sound/sof.h b/include/sound/sof.h index f3e716c8ce1c..9aa055289dcc 100644 --- a/include/sound/sof.h +++ b/include/sound/sof.h @@ -66,6 +66,8 @@ struct sof_dev_desc { /* alternate list of machines using this configuration */ struct snd_soc_acpi_mach *alt_machines; + bool use_acpi_target_states; + /* Platform resource indexes in BAR / ACPI resources. */ /* Must set to -1 if not used - add new items to end */ int resindex_lpe_base; diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index a5f7c7f024a1..c83fb6255961 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -305,15 +305,17 @@ EXPORT_SYMBOL(snd_sof_suspend); int snd_sof_prepare(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); + const struct sof_dev_desc *desc = sdev->pdata->desc; + + /* will suspend to S3 by default */ + sdev->system_suspend_target = SOF_SUSPEND_S3; + + if (!desc->use_acpi_target_states) + return 0; #if defined(CONFIG_ACPI) if (acpi_target_system_state() == ACPI_STATE_S0) sdev->system_suspend_target = SOF_SUSPEND_S0IX; - else - sdev->system_suspend_target = SOF_SUSPEND_S3; -#else - /* will suspend to S3 by default */ - sdev->system_suspend_target = SOF_SUSPEND_S3; #endif return 0; diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 8c53f6935417..8f62e3487dc1 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -77,6 +77,7 @@ static const struct dmi_system_id community_key_platforms[] = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) static const struct sof_dev_desc bxt_desc = { .machines = snd_soc_acpi_intel_bxt_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -94,6 +95,7 @@ static const struct sof_dev_desc bxt_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) static const struct sof_dev_desc glk_desc = { .machines = snd_soc_acpi_intel_glk_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -139,6 +141,7 @@ static const struct sof_dev_desc tng_desc = { static const struct sof_dev_desc cnl_desc = { .machines = snd_soc_acpi_intel_cnl_machines, .alt_machines = snd_soc_acpi_intel_cnl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -157,6 +160,7 @@ static const struct sof_dev_desc cnl_desc = { static const struct sof_dev_desc cfl_desc = { .machines = snd_soc_acpi_intel_cfl_machines, .alt_machines = snd_soc_acpi_intel_cfl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -175,6 +179,7 @@ static const struct sof_dev_desc cfl_desc = { static const struct sof_dev_desc cml_desc = { .machines = snd_soc_acpi_intel_cml_machines, .alt_machines = snd_soc_acpi_intel_cml_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -193,6 +198,7 @@ static const struct sof_dev_desc cml_desc = { static const struct sof_dev_desc icl_desc = { .machines = snd_soc_acpi_intel_icl_machines, .alt_machines = snd_soc_acpi_intel_icl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -211,6 +217,7 @@ static const struct sof_dev_desc icl_desc = { static const struct sof_dev_desc tgl_desc = { .machines = snd_soc_acpi_intel_tgl_machines, .alt_machines = snd_soc_acpi_intel_tgl_sdw_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -244,6 +251,7 @@ static const struct sof_dev_desc tglh_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) static const struct sof_dev_desc ehl_desc = { .machines = snd_soc_acpi_intel_ehl_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, @@ -261,6 +269,7 @@ static const struct sof_dev_desc ehl_desc = { #if IS_ENABLED(CONFIG_SND_SOC_SOF_JASPERLAKE) static const struct sof_dev_desc jsl_desc = { .machines = snd_soc_acpi_intel_jsl_machines, + .use_acpi_target_states = true, .resindex_lpe_base = 0, .resindex_pcicfg_base = -1, .resindex_imr_base = -1, -- 2.27.0

2 1

[PATCH v3 1/2] dt-bindings: tlv320adcx140: Add GPIO config and drive config
by Camel Guo 21 Sep '20

21 Sep '20

From: Camel Guo <camelg(a)axis.com> Add properties for configuring the General Purpose Input Output (GPIO). There are 2 settings for GPIO, configuration and the output drive type. Signed-off-by: Camel Guo <camelg(a)axis.com> Acked-by: Dan Murphy <dmurphy(a)ti.com> --- v3: - Fix typo - Add Acked-By from Dan .../bindings/sound/tlv320adcx140.yaml | 44 +++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index f578f17f3e04..2f95ccde4dc3 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -134,6 +134,49 @@ patternProperties: 4d - Drive weak low and active high 5d - Drive Hi-Z and active high + ti,gpio-config: + description: | + Defines the configuration and output drive for the General Purpose + Input and Output pin (GPIO1). Its value is a pair, the first value is for + the configuration type and the second value is for the output drive + type. The array is defined as <GPIO1_CFG GPIO1_DRV> + + configuration for the GPIO pin can be one of the following: + 0 - disabled + 1 - GPIO1 is configured as a general-purpose output (GPO) + 2 - (default) GPIO1 is configured as a device interrupt output (IRQ) + 3 - GPIO1 is configured as a secondary ASI output (SDOUT2) + 4 - GPIO1 is configured as a PDM clock output (PDMCLK) + 8 - GPIO1 is configured as an input to control when MICBIAS turns on or + off (MICBIAS_EN) + 9 - GPIO1 is configured as a general-purpose input (GPI) + 10 - GPIO1 is configured as a master clock input (MCLK) + 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN) + 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2 + (PDMDIN1) + 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4 + (PDMDIN2) + 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6 + (PDMDIN3) + 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8 + (PDMDIN4) + + output drive type for the GPIO pin can be one of the following: + 0 - Hi-Z output + 1 - Drive active low and active high + 2 - (default) Drive active low and weak high + 3 - Drive active low and Hi-Z + 4 - Drive weak low and active high + 5 - Drive Hi-Z and active high + + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32-array + - minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + required: - compatible - reg @@ -150,6 +193,7 @@ examples: ti,mic-bias-source = <6>; ti,pdm-edge-select = <0 1 0 1>; ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; ti,gpo-config-1 = <0 0>; ti,gpo-config-2 = <0 0>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; -- 2.20.1

3 4

[PATCH v5 1/2] dt-bindings: tlv320adcx140: Add GPIO config and drive config
by Camel Guo 21 Sep '20

21 Sep '20

From: Camel Guo <camelg(a)axis.com> Add properties for configuring the General Purpose Input Output (GPIO). There are 2 settings for GPIO, configuration and the output drive type. Signed-off-by: Camel Guo <camelg(a)axis.com> Acked-by: Dan Murphy <dmurphy(a)ti.com> --- v4: - Rebase v3: - Fix typo - Add Acked-By from Dan .../bindings/sound/tlv320adcx140.yaml | 44 +++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index e79f8d1891e4..798b366fe6d1 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -140,6 +140,49 @@ patternProperties: 4d - Drive weak low and active high 5d - Drive Hi-Z and active high + ti,gpio-config: + description: | + Defines the configuration and output drive for the General Purpose + Input and Output pin (GPIO1). Its value is a pair, the first value is for + the configuration type and the second value is for the output drive + type. The array is defined as <GPIO1_CFG GPIO1_DRV> + + configuration for the GPIO pin can be one of the following: + 0 - disabled + 1 - GPIO1 is configured as a general-purpose output (GPO) + 2 - (default) GPIO1 is configured as a device interrupt output (IRQ) + 3 - GPIO1 is configured as a secondary ASI output (SDOUT2) + 4 - GPIO1 is configured as a PDM clock output (PDMCLK) + 8 - GPIO1 is configured as an input to control when MICBIAS turns on or + off (MICBIAS_EN) + 9 - GPIO1 is configured as a general-purpose input (GPI) + 10 - GPIO1 is configured as a master clock input (MCLK) + 11 - GPIO1 is configured as an ASI input for daisy-chain (SDIN) + 12 - GPIO1 is configured as a PDM data input for channel 1 and channel 2 + (PDMDIN1) + 13 - GPIO1 is configured as a PDM data input for channel 3 and channel 4 + (PDMDIN2) + 14 - GPIO1 is configured as a PDM data input for channel 5 and channel 6 + (PDMDIN3) + 15 - GPIO1 is configured as a PDM data input for channel 7 and channel 8 + (PDMDIN4) + + output drive type for the GPIO pin can be one of the following: + 0 - Hi-Z output + 1 - Drive active low and active high + 2 - (default) Drive active low and weak high + 3 - Drive active low and Hi-Z + 4 - Drive weak low and active high + 5 - Drive Hi-Z and active high + + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32-array + - minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] + required: - compatible - reg @@ -156,6 +199,7 @@ examples: ti,mic-bias-source = <6>; ti,pdm-edge-select = <0 1 0 1>; ti,gpi-config = <4 5 6 7>; + ti,gpio-config = <10 2>; ti,gpo-config-1 = <0 0>; ti,gpo-config-2 = <0 0>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; -- 2.20.1

2 2

[PATCH] ASoC: SOF: imx: add missing MODULE_LICENSE() for imx-common
by Kai Vehmanen 21 Sep '20

21 Sep '20

Fix build warning: WARNING: modpost: missing MODULE_LICENSE() in sound/soc/sof/imx/imx-common.o Fixes: 18ebffe4d043 ("ASoC: SOF: imx: Add debug support for imx platforms") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/imx/imx-common.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/imx/imx-common.c b/sound/soc/sof/imx/imx-common.c index 63d8a5d7bc44..5fee637834c2 100644 --- a/sound/soc/sof/imx/imx-common.c +++ b/sound/soc/sof/imx/imx-common.c @@ -4,6 +4,7 @@ // // Common helpers for the audio DSP on i.MX8 +#include <linux/module.h> #include <sound/sof/xtensa.h> #include "../ops.h" @@ -70,3 +71,5 @@ void imx8_dump(struct snd_sof_dev *sdev, u32 flags) IMX8_STACK_DUMP_SIZE); } EXPORT_SYMBOL(imx8_dump); + +MODULE_LICENSE("Dual BSD/GPL"); -- 2.27.0

2 1

[PATCH] ASoC: cs42l51: add additional ADC volume controls
by Guillermo Rodríguez 21 Sep '20

21 Sep '20

Add volume controls for: - Analog programmable gain amplifier (PGA) (-3 .. +12 dB) - ADC attenuator (0 .. -96 dB) Signed-off-by: Guillermo Rodríguez <guille.rodriguez(a)gmail.com> --- sound/soc/codecs/cs42l51.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 764f2ef8f59d..097c4e8d9950 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -122,6 +122,9 @@ static const char *chan_mix[] = { "R L", }; +static const DECLARE_TLV_DB_SCALE(pga_tlv, -300, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_att_tlv, -9600, 100, 0); + static SOC_ENUM_SINGLE_EXT_DECL(cs42l51_chan_mix, chan_mix); static const struct snd_kcontrol_new cs42l51_snd_controls[] = { @@ -138,6 +141,12 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), + SOC_DOUBLE_R_SX_TLV("ADC Attenuator Volume", + CS42L51_ADCA_ATT, CS42L51_ADCB_ATT, + 0, 0xA0, 96, adc_att_tlv), + SOC_DOUBLE_R_SX_TLV("PGA Volume", + CS42L51_ALC_PGA_CTL, CS42L51_ALC_PGB_CTL, + 0, 0x1A, 30, pga_tlv), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), SOC_SINGLE("Auto-Mute Switch", CS42L51_DAC_CTL, 2, 1, 0), SOC_SINGLE("Soft Ramp Switch", CS42L51_DAC_CTL, 1, 1, 0), -- 2.26.0

3 2

[PATCH 1/2] dt-bindings: tas2562: Add TAS2564 to binding
by Dan Murphy 21 Sep '20

21 Sep '20

Add the TAS2564 compatible and data sheet link to the binding. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tas2562.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml index c3b7e19a0d44..d3ceda356aaa 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.yaml +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -19,12 +19,14 @@ description: | Specifications about the audio amplifier can be found at: https://www.ti.com/lit/gpn/tas2562 https://www.ti.com/lit/gpn/tas2563 + https://www.ti.com/lit/gpn/tas2564 properties: compatible: enum: - ti,tas2562 - ti,tas2563 + - ti,tas2564 reg: maxItems: 1 -- 2.28.0

2 2

[PATCH] ASoC: SOF: topology: fix the process being scheduled on core0 always
by Kai Vehmanen 21 Sep '20

21 Sep '20

From: Keyon Jie <yang.jie(a)linux.intel.com> In commit 783898ce68de ("ASoC: SOF: append extended data to sof_ipc_comp_process") the process components are set to run on the fixed core 0, this break us from scheduling components on any other DSP core. Since we can get the DSP core index from swidget->core, it is duplicated to pass the extra 'core' argument for those sof_widget_load_xx() functions. Here removes the duplicate 'core' argument and get component core from swidget->core directly to fix the issue mentioned above. Fixes: 783898ce68de ("ASoC: SOF: append extended data to sof_ipc_comp_process") Signed-off-by: Keyon Jie <yang.jie(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/topology.c | 82 +++++++++++++++++----------------------- 1 file changed, 35 insertions(+), 47 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index eaa1122d5a68..69313fbdb636 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1465,13 +1465,11 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, * @ipc_size: IPC payload size that will be updated depending on valid * extended data. * @index: ID of the pipeline the component belongs to - * @core: index of the DSP core that the component should run on * * Return: The pointer to the new allocated component, NULL if failed. */ static struct sof_ipc_comp *sof_comp_alloc(struct snd_sof_widget *swidget, - size_t *ipc_size, int index, - int core) + size_t *ipc_size, int index) { u8 nil_uuid[SOF_UUID_SIZE] = {0}; struct sof_ipc_comp *comp; @@ -1490,7 +1488,7 @@ static struct sof_ipc_comp *sof_comp_alloc(struct snd_sof_widget *swidget, comp->hdr.cmd = SOF_IPC_GLB_TPLG_MSG | SOF_IPC_TPLG_COMP_NEW; comp->id = swidget->comp_id; comp->pipeline_id = index; - comp->core = core; + comp->core = swidget->core; /* handle the extended data if needed */ if (total_size > *ipc_size) { @@ -1505,7 +1503,7 @@ static struct sof_ipc_comp *sof_comp_alloc(struct snd_sof_widget *swidget, } static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r, struct snd_sof_dai *dai) @@ -1517,7 +1515,7 @@ static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, int ret; comp_dai = (struct sof_ipc_comp_dai *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!comp_dai) return -ENOMEM; @@ -1571,7 +1569,7 @@ static int sof_widget_load_dai(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_buffer(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1590,7 +1588,7 @@ static int sof_widget_load_buffer(struct snd_soc_component *scomp, int index, buffer->comp.id = swidget->comp_id; buffer->comp.type = SOF_COMP_BUFFER; buffer->comp.pipeline_id = index; - buffer->comp.core = core; + buffer->comp.core = swidget->core; ret = sof_parse_tokens(scomp, buffer, buffer_tokens, ARRAY_SIZE(buffer_tokens), private->array, @@ -1642,7 +1640,7 @@ static int spcm_bind(struct snd_soc_component *scomp, struct snd_sof_pcm *spcm, */ static int sof_widget_load_pcm(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, enum sof_ipc_stream_direction dir, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) @@ -1654,7 +1652,7 @@ static int sof_widget_load_pcm(struct snd_soc_component *scomp, int index, int ret; host = (struct sof_ipc_comp_host *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!host) return -ENOMEM; @@ -1779,7 +1777,7 @@ static int sof_widget_load_pipeline(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1790,7 +1788,7 @@ static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, int ret; mixer = (struct sof_ipc_comp_mixer *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!mixer) return -ENOMEM; @@ -1824,7 +1822,7 @@ static int sof_widget_load_mixer(struct snd_soc_component *scomp, int index, * Mux topology */ static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1835,7 +1833,7 @@ static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, int ret; mux = (struct sof_ipc_comp_mux *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!mux) return -ENOMEM; @@ -1870,7 +1868,7 @@ static int sof_widget_load_mux(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1884,7 +1882,7 @@ static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, int ret; volume = (struct sof_ipc_comp_volume *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!volume) return -ENOMEM; @@ -1946,7 +1944,7 @@ static int sof_widget_load_pga(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_src(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -1957,7 +1955,7 @@ static int sof_widget_load_src(struct snd_soc_component *scomp, int index, int ret; src = (struct sof_ipc_comp_src *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!src) return -ENOMEM; @@ -2003,7 +2001,7 @@ static int sof_widget_load_src(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_asrc(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -2014,7 +2012,7 @@ static int sof_widget_load_asrc(struct snd_soc_component *scomp, int index, int ret; asrc = (struct sof_ipc_comp_asrc *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!asrc) return -ENOMEM; @@ -2062,7 +2060,7 @@ static int sof_widget_load_asrc(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -2073,7 +2071,7 @@ static int sof_widget_load_siggen(struct snd_soc_component *scomp, int index, int ret; tone = (struct sof_ipc_comp_tone *) - sof_comp_alloc(swidget, &ipc_size, index, core); + sof_comp_alloc(swidget, &ipc_size, index); if (!tone) return -ENOMEM; @@ -2229,7 +2227,7 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, } process = (struct sof_ipc_comp_process *) - sof_comp_alloc(swidget, &ipc_size, index, 0); + sof_comp_alloc(swidget, &ipc_size, index); if (!process) { ret = -ENOMEM; goto out; @@ -2307,7 +2305,7 @@ static int sof_process_load(struct snd_soc_component *scomp, int index, */ static int sof_widget_load_process(struct snd_soc_component *scomp, int index, - struct snd_sof_widget *swidget, int core, + struct snd_sof_widget *swidget, struct snd_soc_tplg_dapm_widget *tw, struct sof_ipc_comp_reply *r) { @@ -2322,7 +2320,7 @@ static int sof_widget_load_process(struct snd_soc_component *scomp, int index, } memset(&config, 0, sizeof(config)); - config.comp.core = core; + config.comp.core = swidget->core; /* get the process token */ ret = sof_parse_tokens(scomp, &config, process_tokens, @@ -2450,8 +2448,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, return -ENOMEM; } - ret = sof_widget_load_dai(scomp, index, swidget, comp.core, - tw, &reply, dai); + ret = sof_widget_load_dai(scomp, index, swidget, tw, &reply, dai); if (ret == 0) { sof_connect_dai_widget(scomp, w, tw, dai); list_add(&dai->list, &sdev->dai_list); @@ -2461,12 +2458,10 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, } break; case snd_soc_dapm_mixer: - ret = sof_widget_load_mixer(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_mixer(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_pga: - ret = sof_widget_load_pga(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_pga(scomp, index, swidget, tw, &reply); /* Find scontrol for this pga and set readback offset*/ list_for_each_entry(scontrol, &sdev->kcontrol_list, list) { if (scontrol->comp_id == swidget->comp_id) { @@ -2476,41 +2471,34 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, } break; case snd_soc_dapm_buffer: - ret = sof_widget_load_buffer(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_buffer(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_scheduler: - ret = sof_widget_load_pipeline(scomp, index, swidget, - tw, &reply); + ret = sof_widget_load_pipeline(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_aif_out: - ret = sof_widget_load_pcm(scomp, index, swidget, comp.core, + ret = sof_widget_load_pcm(scomp, index, swidget, SOF_IPC_STREAM_CAPTURE, tw, &reply); break; case snd_soc_dapm_aif_in: - ret = sof_widget_load_pcm(scomp, index, swidget, comp.core, + ret = sof_widget_load_pcm(scomp, index, swidget, SOF_IPC_STREAM_PLAYBACK, tw, &reply); break; case snd_soc_dapm_src: - ret = sof_widget_load_src(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_src(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_asrc: - ret = sof_widget_load_asrc(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_asrc(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_siggen: - ret = sof_widget_load_siggen(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_siggen(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_effect: - ret = sof_widget_load_process(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_process(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_mux: case snd_soc_dapm_demux: - ret = sof_widget_load_mux(scomp, index, swidget, comp.core, - tw, &reply); + ret = sof_widget_load_mux(scomp, index, swidget, tw, &reply); break; case snd_soc_dapm_switch: case snd_soc_dapm_dai_link: -- 2.27.0

2 1

[PATCH] ALSA: usb-audio: Add mixer support for Pioneer DJ DJM-250MK2
by František Kučera 21 Sep '20

21 Sep '20

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows mapping playback and capture channels to available sources. Configures the card through USB commands. --- sound/usb/mixer_quirks.c | 194 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 194 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 199cdbfdc761..f2aeb5ac3611 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2602,6 +2602,197 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) return 0; } +/* + * Pioneer DJ DJM-250MK2 and maybe other DJM models + * + * For playback, no duplicate mapping should be set. + * There are three mixer stereo channels (CH1, CH2, AUX) and three stereo sources (PLAYBACK 1-2, PLAYBACK 3-4, PLAYBACK 5-6). + * Each channel should be mapped just once to one source. + * If mapped multiple times, only one source will play on given channel (sources are not mixed together). + * + * For recording, duplicate mapping is OK. We will get the same signal multiple times. + * + * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN. + * + * See also notes in the quirks-table.h file. + */ + +typedef struct { + const u16 wIndex; + const u16 wValue; + const char *name; +} snd_pioneer_djm_option; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_LEVEL[] = { + { .name = "-5 dB", .wValue = 0x0300, .wIndex = 0x8003 }, + { .name = "-10 dB", .wValue = 0x0200, .wIndex = 0x8003 }, + { .name = "-15 dB", .wValue = 0x0100, .wIndex = 0x8003 }, + { .name = "-19 dB", .wValue = 0x0000, .wIndex = 0x8003 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH12[] = { + { .name = "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 }, + { .name = "CH1 Control Tone LINE", .wValue = 0x0100, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0106, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0107, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0108, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0109, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x010d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x010a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH34[] = { + { .name = "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 }, + { .name = "CH2 Control Tone LINE", .wValue = 0x0200, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0206, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0207, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0208, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0209, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x020d, .wIndex = 0x8002 }, + { .name = "REC OUT", .wValue = 0x020a, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_CAPTURE_CH56[] = { + { .name = "REC OUT", .wValue = 0x030a, .wIndex = 0x8002 }, + { .name = "Post CH1 Fader", .wValue = 0x0311, .wIndex = 0x8002 }, + { .name = "Post CH2 Fader", .wValue = 0x0312, .wIndex = 0x8002 }, + { .name = "Cross Fader A", .wValue = 0x0307, .wIndex = 0x8002 }, + { .name = "Cross Fader B", .wValue = 0x0308, .wIndex = 0x8002 }, + { .name = "MIC", .wValue = 0x0309, .wIndex = 0x8002 }, + { .name = "AUX", .wValue = 0x030d, .wIndex = 0x8002 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_12[] = { + { .name = "CH1", .wValue = 0x0100, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0101, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0104, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_34[] = { + { .name = "CH1", .wValue = 0x0200, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0201, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0204, .wIndex = 0x8016 } +}; + +static const snd_pioneer_djm_option SND_PIONEER_DJM_OPTIONS_PLAYBACK_56[] = { + { .name = "CH1", .wValue = 0x0300, .wIndex = 0x8016 }, + { .name = "CH2", .wValue = 0x0301, .wIndex = 0x8016 }, + { .name = "AUX", .wValue = 0x0304, .wIndex = 0x8016 } +}; + +typedef struct { + const char *name; + const snd_pioneer_djm_option *options; + const size_t count; + const u16 default_value; +} snd_pioneer_djm_option_group; + +#define SND_PIONEER_DJM_OPTION_GROUP_ITEM(_name, suffix, _default_value) \ + { .name = _name, .options = SND_PIONEER_DJM_OPTIONS_##suffix, .count = ARRAY_SIZE(SND_PIONEER_DJM_OPTIONS_##suffix), .default_value = _default_value } + +static const snd_pioneer_djm_option_group SND_PIONEER_DJM_OPTION_GROUPS[] = { + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Master Capture Level Capture Switch", CAPTURE_LEVEL, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 1-2 Capture Switch", CAPTURE_CH12, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 3-4 Capture Switch", CAPTURE_CH34, 2), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Capture 5-6 Capture Switch", CAPTURE_CH56, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 1-2 Playback Switch", PLAYBACK_12, 0), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 3-4 Playback Switch", PLAYBACK_34, 1), + SND_PIONEER_DJM_OPTION_GROUP_ITEM("Playback 5-6 Playback Switch", PLAYBACK_56, 2) +}; + +// layout of the kcontrol->private_value: +#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff +#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000 +#define SND_PIONEER_DJM_GROUP_SHIFT 16 + +static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *info) +{ + u16 group = kcontrol->private_value >> SND_PIONEER_DJM_GROUP_SHIFT; + size_t count; + const char *name; + if (group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS)) { + count = SND_PIONEER_DJM_OPTION_GROUPS[group].count; + if (info->value.enumerated.item >= count) info->value.enumerated.item = count - 1; + name = SND_PIONEER_DJM_OPTION_GROUPS[group].options[info->value.enumerated.item].name; + strlcpy(info->value.enumerated.name, name, sizeof (info->value.enumerated.name)); + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = count; + return 0; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value) +{ + int err; + if (group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS) && value < SND_PIONEER_DJM_OPTION_GROUPS[group].count) { + err = snd_usb_lock_shutdown(mixer->chip); + if (err) return err; + + err = snd_usb_ctl_msg( + mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0), + USB_REQ_SET_FEATURE, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wValue, + SND_PIONEER_DJM_OPTION_GROUPS[group].options[value].wIndex, + NULL, 0); + + snd_usb_unlock_shutdown(mixer->chip); + return err; + } else { + return -EINVAL; + } +} + +static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = kcontrol->private_value & SND_PIONEER_DJM_VALUE_MASK; + return 0; +} + +static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kcontrol); + struct usb_mixer_interface *mixer = list->mixer; + + u16 group = (kcontrol->private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = ucontrol->value.enumerated.item[0]; + kcontrol->private_value = group << SND_PIONEER_DJM_GROUP_SHIFT | value; + + return snd_pioneer_djm_controls_update(mixer, group, value); +} + +static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list) +{ + u16 group = (list->kctl->private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT; + u16 value = (list->kctl->private_value & SND_PIONEER_DJM_VALUE_MASK); + return snd_pioneer_djm_controls_update(list->mixer, group, value); +} + +static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer) +{ + int err, group; + for (group = 0; group < ARRAY_SIZE(SND_PIONEER_DJM_OPTION_GROUPS); group++ ) { + struct snd_kcontrol_new knew = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .index = 0, + .info = snd_pioneer_djm_controls_info, + .get = snd_pioneer_djm_controls_get, + .put = snd_pioneer_djm_controls_put, + .name = SND_PIONEER_DJM_OPTION_GROUPS[group].name, + .private_value = group << SND_PIONEER_DJM_GROUP_SHIFT | SND_PIONEER_DJM_OPTION_GROUPS[group].default_value + }; + err = snd_pioneer_djm_controls_update(mixer, group, SND_PIONEER_DJM_OPTION_GROUPS[group].default_value); + if (err) return err; + err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume, &knew, NULL); + if (err) return err; + } + return 0; +} + int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) { int err = 0; @@ -2706,6 +2897,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */ err = snd_bbfpro_controls_create(mixer); break; + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ + err = snd_pioneer_djm_controls_create(mixer); + break; } return err; -- 2.20.1

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[PATCH 0/4] ALSA: hda - add Intel DG1 support
by Kai Vehmanen 21 Sep '20

21 Sep '20

Series adding support for HDMI/DP audio for Intel DG1. Kai Vehmanen (4): ALSA: hda - add Intel DG1 PCI and HDMI ids ALSA: hda - controller is in GPU on the DG1 ALSA: hda - handle multiple i915 device instances ALSA: hda - fix CONTROLLER_IN_GPU macro name sound/hda/hdac_i915.c | 48 ++++++++++++++++++++++++++++++++++---- sound/pci/hda/hda_intel.c | 6 ++++- sound/pci/hda/patch_hdmi.c | 1 + 3 files changed, 50 insertions(+), 5 deletions(-) -- 2.27.0

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[PATCH AUTOSEL 4.19 1/9] ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811
by Sasha Levin 21 Sep '20

21 Sep '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit 811c5494436789e7149487c06e0602b507ce274b ] The WM8994_MICBIAS register is not available in the WM1811 CODEC so skip initialization of that register for that device. This suppresses an error during boot: "wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec" Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20200827173357.31891-1-s.nawrocki@samsung.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm_hubs.c | 3 +++ sound/soc/codecs/wm_hubs.h | 1 + 3 files changed, 6 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 01acb8da2f48e..cd089b4143029 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4051,11 +4051,13 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994->hubs.dcs_readback_mode = 2; break; } + wm8994->hubs.micd_scthr = true; break; case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.micd_scthr = true; switch (control->revision) { case 0: diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index fed6ea9b019f7..da7fa6f5459e6 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1227,6 +1227,9 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + if (!hubs->micd_scthr) + return 0; + snd_soc_component_update_bits(component, WM8993_MICBIAS, WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | WM8993_MICB1_LVL | WM8993_MICB2_LVL, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index ee339ad8514d1..1433d73e09bf8 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -31,6 +31,7 @@ struct wm_hubs_data { int hp_startup_mode; int series_startup; int no_series_update; + bool micd_scthr; bool no_cache_dac_hp_direct; struct list_head dcs_cache; -- 2.25.1

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