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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

2 1

[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

2 11

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 */ >

1 0

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)

1 0

[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.

5 14

[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

2 8

[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

2 1

[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

2 1

[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

2 5

[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

1 2

[PATCH AUTOSEL 5.4 05/15] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN Converter9 2-in-1
by Sasha Levin 21 Sep '20

21 Sep '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 6a0137101f47301fff2da6ba4b9048383d569909 ] The MPMAN Converter9 2-in-1 almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200901080623.4987-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index f7964d1ec486f..6012367f6fe48 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* MPMAN Converter 9, similar hw as the I.T.Works TW891 2-in-1 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_MATCH(DMI_PRODUCT_NAME, "Converter9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MPMAN MPWIN895CL */ .matches = { -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 04/15] ASoC: wm8994: Ensure the device is resumed in wm89xx_mic_detect functions
by Sasha Levin 21 Sep '20

21 Sep '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit f5a2cda4f1db89776b64c4f0f2c2ac609527ac70 ] When the wm8958_mic_detect, wm8994_mic_detect functions get called from the machine driver, e.g. from the card's late_probe() callback, the CODEC device may be PM runtime suspended and any regmap writes have no effect. Add PM runtime calls to these functions to ensure the device registers are updated as expected. This suppresses an error during boot "wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec" caused by the regmap access error due to the cache_only flag being set. 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/20200827173357.31891-2-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 | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 64635f9cdae65..6dbab3fc6537e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3372,6 +3372,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * return -EINVAL; } + pm_runtime_get_sync(component->dev); + switch (micbias) { case 1: micdet = &wm8994->micdet[0]; @@ -3419,6 +3421,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * snd_soc_dapm_sync(dapm); + pm_runtime_put(component->dev); + return 0; } EXPORT_SYMBOL_GPL(wm8994_mic_detect); @@ -3786,6 +3790,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * return -EINVAL; } + pm_runtime_get_sync(component->dev); + if (jack) { snd_soc_dapm_force_enable_pin(dapm, "CLK_SYS"); snd_soc_dapm_sync(dapm); @@ -3854,6 +3860,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * snd_soc_dapm_sync(dapm); } + pm_runtime_put(component->dev); + return 0; } EXPORT_SYMBOL_GPL(wm8958_mic_detect); -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 03/15] 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 d5fb7f5dd551c..64635f9cdae65 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4047,11 +4047,13 @@ 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

1 0

[PATCH AUTOSEL 5.4 02/15] ASoC: pcm3168a: ignore 0 Hz settings
by Sasha Levin 21 Sep '20

21 Sep '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 7ad26d6671db758c959d7e1d100b138a38483612 ] Some sound card try to set 0 Hz as reset, but it is impossible. This patch ignores it to avoid error return. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87a6yjy5sy.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/pcm3168a.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 88b75695fbf7f..b37e5fbbd301a 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -302,6 +302,13 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component); int ret; + /* + * Some sound card sets 0 Hz as reset, + * but it is impossible to set. Ignore it here + */ + if (freq == 0) + return 0; + if (freq > PCM3168A_MAX_SYSCLK) return -EINVAL; -- 2.25.1

1 0

[PATCH AUTOSEL 5.8 05/20] ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN Converter9 2-in-1
by Sasha Levin 21 Sep '20

21 Sep '20

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 6a0137101f47301fff2da6ba4b9048383d569909 ] The MPMAN Converter9 2-in-1 almost fully works with out default settings. The only problem is that it has only 1 speaker so any sounds only playing on the right channel get lost. Add a quirk for this model using the default settings + MONO_SPEAKER. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200901080623.4987-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 1fdb70b9e4788..5f885062145fe 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* MPMAN Converter 9, similar hw as the I.T.Works TW891 2-in-1 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_MATCH(DMI_PRODUCT_NAME, "Converter9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MPMAN MPWIN895CL */ .matches = { -- 2.25.1

1 0

[PATCH AUTOSEL 5.8 04/20] ASoC: wm8994: Ensure the device is resumed in wm89xx_mic_detect functions
by Sasha Levin 21 Sep '20

21 Sep '20

From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> [ Upstream commit f5a2cda4f1db89776b64c4f0f2c2ac609527ac70 ] When the wm8958_mic_detect, wm8994_mic_detect functions get called from the machine driver, e.g. from the card's late_probe() callback, the CODEC device may be PM runtime suspended and any regmap writes have no effect. Add PM runtime calls to these functions to ensure the device registers are updated as expected. This suppresses an error during boot "wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec" caused by the regmap access error due to the cache_only flag being set. 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/20200827173357.31891-2-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 | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c2116836a7203..58f21329d0e99 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3491,6 +3491,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * return -EINVAL; } + pm_runtime_get_sync(component->dev); + switch (micbias) { case 1: micdet = &wm8994->micdet[0]; @@ -3538,6 +3540,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * snd_soc_dapm_sync(dapm); + pm_runtime_put(component->dev); + return 0; } EXPORT_SYMBOL_GPL(wm8994_mic_detect); @@ -3905,6 +3909,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * return -EINVAL; } + pm_runtime_get_sync(component->dev); + if (jack) { snd_soc_dapm_force_enable_pin(dapm, "CLK_SYS"); snd_soc_dapm_sync(dapm); @@ -3973,6 +3979,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * snd_soc_dapm_sync(dapm); } + pm_runtime_put(component->dev); + return 0; } EXPORT_SYMBOL_GPL(wm8958_mic_detect); -- 2.25.1

1 0

[PATCH AUTOSEL 5.8 02/20] ASoC: pcm3168a: ignore 0 Hz settings
by Sasha Levin 21 Sep '20

21 Sep '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit 7ad26d6671db758c959d7e1d100b138a38483612 ] Some sound card try to set 0 Hz as reset, but it is impossible. This patch ignores it to avoid error return. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87a6yjy5sy.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/pcm3168a.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 9711fab296ebc..045c6f8b26bef 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -306,6 +306,13 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component); int ret; + /* + * Some sound card sets 0 Hz as reset, + * but it is impossible to set. Ignore it here + */ + if (freq == 0) + return 0; + if (freq > PCM3168A_MAX_SYSCLK) return -EINVAL; -- 2.25.1

1 0

[PATCH 1/2] ASoC: rt700: wait for the delayed work to finish when the system suspends
by shumingf＠realtek.com 21 Sep '20

21 Sep '20

From: Shuming Fan <shumingf(a)realtek.com> To avoid the IO error, we need to cancel the delayed work and wait for it to finish. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt700-sdw.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index ead4918bbf90..c7deb4e4fcf1 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -490,6 +490,9 @@ static int __maybe_unused rt700_dev_suspend(struct device *dev) if (!rt700->hw_init) return 0; + cancel_delayed_work_sync(&rt700->jack_detect_work); + cancel_delayed_work_sync(&rt700->jack_btn_check_work); + regcache_cache_only(rt700->regmap, true); return 0; -- 2.28.0

2 1

[PATCH 2/2] ASoC: rt711: wait for the delayed work to finish when the system suspends
by shumingf＠realtek.com 21 Sep '20

21 Sep '20

From: Shuming Fan <shumingf(a)realtek.com> To avoid the IO error, we need to cancel the delayed work and wait for it to finish. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdw.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 28d663673320..3a8ca600d1cf 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -491,6 +491,10 @@ static int __maybe_unused rt711_dev_suspend(struct device *dev) if (!rt711->hw_init) return 0; + cancel_delayed_work_sync(&rt711->jack_detect_work); + cancel_delayed_work_sync(&rt711->jack_btn_check_work); + cancel_work_sync(&rt711->calibration_work); + regcache_cache_only(rt711->regmap, true); return 0; -- 2.28.0

2 1

[PATCH v5 00/13] 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. 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 v5: 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 (13): ASoC: Intel: Add catpt device ASoC: Intel: catpt: Define DSP operations ASoC: Intel: catpt: Firmware loading and context restore ASoC: Intel: catpt: Implement IPC protocol ASoC: Intel: catpt: Add IPC messages ASoC: Intel: catpt: PCM operations 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 | 187 +++++ 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 | 673 +++++++++++++++ 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

4 30

[PATCH] Revert "ALSA: hda - Fix silent audio output and corrupted input on MSI X570-A PRO"
by Takashi Iwai 21 Sep '20

21 Sep '20

This reverts commit 15cbff3fbbc6 ("ALSA: hda - Fix silent audio output and corrupted input on MSI X570-A PRO"). A regression reported by a Fedora user for MSI X570-A PRO mobo. Until the correct solution is found out, let's revert the quirk as a quick workaround. Fixes: 15cbff3fbbc6 ("ALSA: hda - Fix silent audio output and corrupted input on MSI X570-A PRO") Reported-by: Hans de Goede <hdegoede(a)redhat.com> Cc: <stable(a)vger.kernel.org> Cc: Dan Crawford <dnlcrwfrd(a)gmail.com> BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1879277 Link: https://lore.kernel.org/r/7efd2fe5-bf38-7f85-891a-eee3845d1493@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c4d615cf8de1..d4f17b465892 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2475,7 +2475,6 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0x1276, "MSI-GL73", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x1293, "MSI-GP65", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0x7350, "MSI-7350", ALC889_FIXUP_CD), - SND_PCI_QUIRK(0x1462, 0x9c37, "MSI X570-A PRO", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1462, 0xda57, "MSI Z270-Gaming", ALC1220_FIXUP_GB_DUAL_CODECS), SND_PCI_QUIRK_VENDOR(0x1462, "MSI", ALC882_FIXUP_GPIO3), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", ALC882_FIXUP_ABIT_AW9D_MAX), -- 2.16.4

1 0

[PATCH v2 1/1] ALSA: hda: Refactor calculating SDnFMT according to specification
by Pawel Harlozinski 21 Sep '20

21 Sep '20

Fix setting SDnFMT based on 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. Change-Id: I97771b16da14e85b7f35372f5dfc87bb13bb5ce0 Signed-off-by: Pawel Harlozinski <pawel.harlozinski(a)linux.intel.com> --- 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

3 2

PROBLEM: A page fault occurs when the sound driver is loaded (ThinkPad X13/T14)
by Philipp Leskovitz 21 Sep '20

21 Sep '20

[1.] One line summary of the problem: A page fault occurs when the sound driver is loaded. [2.] Full description of the problem/report: A page fault occurs when the sound driver for Intel Commet-Lake platform is loaded. This happens e.g. on Lenovo ThinkPad X13 and T14 at startup and version 1.4.3 of intel-sof-firmware binaries (https://github.com/thesofproject/sof-bin/tree/stable-v1.4.3). Same result with version 1.5.1. [3.] Keywords (i.e., modules, networking, kernel): [4.] Kernel information [4.1.] Kernel version (from /proc/version): >5.9-rc5 commit-id: fc4f28bb3daf3265d6bc5f73b497306985bb23ab [4.2.] Kernel .config file: See attachment for full configuration, the important part is: ... CONFIG_SND_SOC=y CONFIG_SND_SOC_INTEL_SST_TOPLEVEL=y CONFIG_SND_SOC_INTEL_CML_LP=y CONFIG_SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC=y CONFIG_SND_SOC_INTEL_MACH=y CONFIG_SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH=y CONFIG_SND_SOC_SOF_TOPLEVEL=y CONFIG_SND_SOC_SOF_PCI=y CONFIG_SND_SOC_SOF_INTEL_TOPLEVEL=y CONFIG_SND_SOC_SOF_COMETLAKE_LP_SUPPORT=y CONFIG_SND_SOC_SOF_HDA_LINK=y CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC=y CONFIG_SND_SOC_SOF_HDA_ALWAYS_ENABLE_DMI_L1=y ... [5.] Most recent kernel version which did not have the bug: - Ubuntu 20.04.1 works fine (5.4.0-45) - Kernel 5.4.61 with some backports from 5.7 [6.] Output of Oops.. message (if applicable) with symbolic information resolved (see Documentation/admin-guide/oops-tracing.rst) See attachment for full log. ... [ 6.380009] snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040380 [ 6.380536] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 6.381915] usbcore: registered new interface driver snd-usb-audio [ 6.382525] sof-audio-pci 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040380 [ 6.383090] sof-audio-pci 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 6.386039] sof-audio-pci 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040380 [ 6.386862] usb 1-1: New USB device found, idVendor=058f, idProduct=9540, bcdDevice= 1.20 [ 6.386959] IPv4 over IPsec tunneling driver [ 6.387207] sof-audio-pci 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops) [ 6.387747] usb 1-1: New USB device strings: Mfr=1, Product=2, SerialNumber=0 [ 6.389474] usb 1-1: Product: EMV Smartcard Reader [ 6.390069] usb 1-1: Manufacturer: Generic [ 6.390892] Initializing XFRM netlink socket [ 6.392858] NET: Registered protocol family 10 [ 6.395179] sof-audio-pci 0000:00:1f.3: use msi interrupt mode [ 6.396219] Segment Routing with IPv6 [ 6.399097] sit: IPv6, IPv4 and MPLS over IPv4 tunneling driver [ 6.402236] NET: Registered protocol family 17 [ 6.405170] NET: Registered protocol family 15 [ 6.406068] 9pnet: Installing 9P2000 support [ 6.408354] microcode: sig=0x806ec, pf=0x4, revision=0xd6 [ 6.409081] microcode: Microcode Update Driver: v2.2. [ 6.409083] IPI shorthand broadcast: enabled [ 6.410374] sched_clock: Marking stable (6404500108, 5227507)->(6418014859, -8287244) [ 6.411127] Loading compiled-in X.509 certificates [ 6.412635] HDMI HDA Codec ehdaudio0D2: Max dais supported: 3 [ 6.413502] sof-audio-pci 0000:00:1f.3: hda codecs found, mask 5 [ 6.414129] sof-audio-pci 0000:00:1f.3: using HDA machine driver skl_hda_dsp_generic now [ 6.414714] sof-audio-pci 0000:00:1f.3: DMICs detected in NHLT tables: 2 [ 6.416290] Loaded X.509 cert 'Build time autogenerated kernel key: b521b2c060c3f56dd5ab1fbfb34c05c2916a8c0c' [ 6.416931] pstore: Using crash dump compression: deflate [ 6.423238] PM: Magic number: 4:125:733 [ 6.423900] hwmon hwmon0: hash matches [ 6.424597] cfg80211: Loading compiled-in X.509 certificates for regulatory database [ 6.429162] cfg80211: Loaded X.509 cert 'sforshee: 00b28ddf47aef9cea7' [ 6.434042] ALSA device list: [ 6.434894] No soundcards found. [ 6.439211] Freeing unused kernel image (initmem) memory: 1572K [ 6.445948] Write protecting the kernel read-only data: 24576k [ 6.448072] Freeing unused kernel image (text/rodata gap) memory: 2044K [ 6.448904] Freeing unused kernel image (rodata/data gap) memory: 368K [ 6.449569] Run /init as init process [ 6.450203] with arguments: [ 6.450203] /init [ 6.450204] with environment: [ 6.450204] HOME=/ [ 6.450204] TERM=linux [ 6.450205] BOOT_IMAGE=/isolinux/bzImage [ 6.450205] intel_iommu=on,igfx_off [ 6.487322] usb 2-4: new SuperSpeed Gen 1 USB device number 2 using xhci_hcd [ 6.501610] usb 2-4: New USB device found, idVendor=0781, idProduct=5580, bcdDevice= 0.10 [ 6.502354] usb 2-4: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 6.503080] usb 2-4: Product: Extreme [ 6.503794] usb 2-4: Manufacturer: SanDisk [ 6.504502] usb 2-4: SerialNumber: AA010121131224373982 [ 6.519956] sof-audio-pci 0000:00:1f.3: Firmware info: version 1:4:3-76052 [ 6.520648] sof-audio-pci 0000:00:1f.3: Firmware: ABI 3:11:0 Kernel ABI 3:16:0 [ 6.521724] sof-audio-pci 0000:00:1f.3: Topology: ABI 3:11:0 Kernel ABI 3:16:0 [ 6.522393] sof-audio-pci 0000:00:1f.3: warning: widget type 7 name iDisp3 Tx not handled [ 6.524941] sof-audio-pci 0000:00:1f.3: warning: widget type 0 name codec0_in not handled [ 6.528203] sof-audio-pci 0000:00:1f.3: warning: widget type 7 name iDisp2 Tx not handled [ 6.533912] sof-audio-pci 0000:00:1f.3: warning: widget type 0 name codec1_in not handled [ 6.534597] sof-audio-pci 0000:00:1f.3: warning: widget type 7 name iDisp1 Tx not handled [ 6.540491] sof-audio-pci 0000:00:1f.3: warning: widget type 1 name codec0_out not handled [ 6.541826] sof-audio-pci 0000:00:1f.3: warning: widget type 7 name Analog CPU Playback not handled [ 6.546622] sof-audio-pci 0000:00:1f.3: warning: widget type 1 name codec1_out not handled [ 6.547556] sof-audio-pci 0000:00:1f.3: warning: widget type 7 name Digital CPU Playback not handled [ 6.548221] sof-audio-pci 0000:00:1f.3: warning: widget type 0 name codec2_in not handled [ 6.548852] sof-audio-pci 0000:00:1f.3: warning: widget type 7 name Alt Analog CPU Playback not handled [ 6.549510] sof-audio-pci 0000:00:1f.3: warning: widget type 1 name codec2_out not handled [ 6.550316] sof-audio-pci 0000:00:1f.3: warning: widget type 0 name Analog CPU Capture not handled [ 6.551743] sof-audio-pci 0000:00:1f.3: warning: widget type 1 name iDisp1_out not handled [ 6.552380] sof-audio-pci 0000:00:1f.3: warning: widget type 0 name Digital CPU Capture not handled [ 6.554251] sof-audio-pci 0000:00:1f.3: warning: widget type 1 name iDisp2_out not handled [ 6.554853] sof-audio-pci 0000:00:1f.3: warning: widget type 0 name Alt Analog CPU Capture not handled [ 6.557362] sof-audio-pci 0000:00:1f.3: warning: widget type 1 name iDisp3_out not handled [ 6.578044] sof-audio-pci 0000:00:1f.3: ASoC: Parent card not yet available, widget card binding deferred [ 6.583577] HDMI HDA Codec ehdaudio0D2: hdac_hdmi_present_sense: disconnect for pin:port 5:0 [ 6.584601] HDMI HDA Codec ehdaudio0D2: hdac_hdmi_present_sense: disconnect for pin:port 6:0 [ 6.585451] HDMI HDA Codec ehdaudio0D2: hdac_hdmi_present_sense: disconnect for pin:port 7:0 [ 6.605984] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC257: line_outs=1 (0x14/0x0/0x0/0x0/0x0) type:speaker [ 6.607282] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 6.608474] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 6.611586] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 6.612904] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 6.615225] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 [ 6.615902] usb 1-8: new high-speed USB device number 3 using xhci_hcd [ 6.667538] skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi3 overwritten [ 6.668264] skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi2 overwritten [ 6.668918] skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: sink widget hifi1 overwritten [ 6.669698] BUG: unable to handle page fault for address: ffffffffffffffe0 [ 6.670415] #PF: supervisor write access in kernel mode [ 6.670993] #PF: error_code(0x0002) - not-present page [ 6.671558] PGD 34180d067 P4D 34180d067 PUD 34180f067 PMD 0 [ 6.672142] Oops: 0002 [#1] SMP NOPTI [ 6.672675] CPU: 2 PID: 126 Comm: kworker/2:2 Not tainted 5.9.0-devel+ #2 [ 6.673225] Hardware name: LENOVO 20S1S19N00/20S1S19N00, BIOS N2XET16W (1.06 ) 05/13/2020 [ 6.673844] Workqueue: events sof_probe_work [ 6.674387] RIP: 0010:hda_dsp_hdmi_build_controls+0x5b/0xff [ 6.674942] Code: 48 8d 98 20 ff ff ff 49 39 c4 74 4c 45 31 ff 49 c7 c6 e0 1a 36 a2 44 89 fe 48 89 ef e8 1e ff ff ff 48 85 c0 0f 84 be 00 00 00 <48> 89 83 c0 00 00 00 44 8b 48 18 44 89 8b bc 00 00 00 0f 1f 44 00 [ 6.675525] RSP: 0000:ffff9e93038e7b08 EFLAGS: 00010286 [ 6.676113] RAX: ffff8ef56180f000 RBX: ffffffffffffff20 RCX: 0000000000000004 [ 6.676705] RDX: 0000000000000004 RSI: ffffffffa26e729f RDI: ffff8ef56180f02d [ 6.677291] RBP: ffffffffa291a240 R08: 0000000000000049 R09: 0000000000000000 [ 6.677889] R10: ffff8ef5635ef024 R11: ffff9e93207d8220 R12: ffff8ef56231dc68 [ 6.678481] R13: ffff8ef56231d828 R14: ffffffffa2361ae0 R15: 0000000000000000 [ 6.679064] FS: 0000000000000000(0000) GS:ffff8ef56f680000(0000) knlGS:0000000000000000 [ 6.679667] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 6.680322] CR2: ffffffffffffffe0 CR3: 0000000820c24003 CR4: 00000000003706e0 [ 6.680968] Call Trace: [ 6.681583] skl_hda_hdmi_jack_init+0x119/0x12b [ 6.682227] ? snprintf+0x49/0x60 [ 6.682844] snd_soc_card_late_probe+0x1e/0x70 [ 6.683461] snd_soc_bind_card+0x83f/0x960 [ 6.684075] devm_snd_soc_register_card+0x43/0x80 [ 6.684676] skl_hda_audio_probe+0x142/0x2e0 [ 6.685265] platform_drv_probe+0x35/0x80 [ 6.685851] really_probe+0x31b/0x430 [ 6.686428] ? driver_allows_async_probing+0x50/0x50 [ 6.687004] driver_probe_device+0xde/0x150 [ 6.687581] ? driver_allows_async_probing+0x50/0x50 [ 6.688162] bus_for_each_drv+0x79/0xc0 [ 6.688736] __device_attach+0xe0/0x160 [ 6.689310] bus_probe_device+0x87/0xa0 [ 6.689885] device_add+0x436/0x810 [ 6.690458] ? _cond_resched+0x15/0x30 [ 6.691032] platform_device_add+0x110/0x240 [ 6.691611] platform_device_register_full+0xc7/0x120 [ 6.692191] sof_machine_register+0x60/0xc0 [ 6.692777] sof_probe_work+0x11c/0x230 [ 6.693357] process_one_work+0x205/0x3d0 [ 6.693938] ? process_one_work+0x3d0/0x3d0 [ 6.694515] worker_thread+0x2d/0x3c0 [ 6.695090] ? process_one_work+0x3d0/0x3d0 [ 6.695664] kthread+0x116/0x130 [ 6.696235] ? kthread_park+0x80/0x80 [ 6.696804] ret_from_fork+0x1f/0x30 [ 6.697372] Modules linked in: [ 6.697936] CR2: ffffffffffffffe0 [7.] A small shell script or example program which triggers the problem (if possible) [8.] Environment [8.1.] Software (add the output of the ver_linux script here) [8.2.] Processor information (from /proc/cpuinfo): [8.3.] Module information (from /proc/modules): [8.4.] Loaded driver and hardware information (/proc/ioports, /proc/iomem) [8.5.] PCI information ('lspci -vvv' as root) [8.6.] SCSI information (from /proc/scsi/scsi) [8.7.] Other information that might be relevant to the problem (please look in /proc and include all information that you think to be relevant): [X.] Other notes, patches, fixes, workarounds:

2 1

[PATCH] ALSA: usb-audio: Add delay quirk for H570e USB headsets
by Joakim Tjernlund 21 Sep '20

21 Sep '20

Needs the same delay as H650e Signed-off-by: Joakim Tjernlund <joakim.tjernlund(a)infinera.com> Cc: stable(a)vger.kernel.org --- sound/usb/quirks.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 75bbdc691243..892296df131d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1678,12 +1678,13 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); - /* Zoom R16/24, Logitech H650e, Jabra 550a, Kingston HyperX needs a tiny - * delay here, otherwise requests like get/set frequency return as - * failed despite actually succeeding. + /* Zoom R16/24, Logitech H650e/H570e, Jabra 550a, Kingston HyperX + * needs a tiny delay here, otherwise requests like get/set + * frequency return as failed despite actually succeeding. */ if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || chip->usb_id == USB_ID(0x046d, 0x0a46) || + chip->usb_id == USB_ID(0x046d, 0x0a56) || chip->usb_id == USB_ID(0x0b0e, 0x0349) || chip->usb_id == USB_ID(0x0951, 0x16ad)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) -- 2.26.2

3 4

[PATCH] ALSA: compat_ioctl: avoid compat_alloc_user_space
by Arnd Bergmann 21 Sep '20

21 Sep '20

Using compat_alloc_user_space() tends to add complexity to the ioctl handling, so I am trying to remove it everywhere. The two callers in sound/core can rewritten to just call the same code that operates on a kernel pointer as the native handler. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/core/control.c | 38 ++++++++++++++++++++++++------------- sound/core/control_compat.c | 14 ++++++-------- sound/core/hwdep.c | 27 ++++++++++++++++---------- sound/core/hwdep_compat.c | 23 +++++++--------------- 4 files changed, 55 insertions(+), 47 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index aa0c0cf182af..e014598142df 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -717,22 +717,19 @@ static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl, } static int snd_ctl_elem_list(struct snd_card *card, - struct snd_ctl_elem_list __user *_list) + struct snd_ctl_elem_list *list) { - struct snd_ctl_elem_list list; struct snd_kcontrol *kctl; struct snd_ctl_elem_id id; unsigned int offset, space, jidx; int err = 0; - if (copy_from_user(&list, _list, sizeof(list))) - return -EFAULT; - offset = list.offset; - space = list.space; + offset = list->offset; + space = list->space; down_read(&card->controls_rwsem); - list.count = card->controls_count; - list.used = 0; + list->count = card->controls_count; + list->used = 0; if (space > 0) { list_for_each_entry(kctl, &card->controls, list) { if (offset >= kctl->count) { @@ -741,12 +738,12 @@ static int snd_ctl_elem_list(struct snd_card *card, } for (jidx = offset; jidx < kctl->count; jidx++) { snd_ctl_build_ioff(&id, kctl, jidx); - if (copy_to_user(list.pids + list.used, &id, + if (copy_to_user(list->pids + list->used, &id, sizeof(id))) { err = -EFAULT; goto out; } - list.used++; + list->used++; if (!--space) goto out; } @@ -755,11 +752,26 @@ static int snd_ctl_elem_list(struct snd_card *card, } out: up_read(&card->controls_rwsem); - if (!err && copy_to_user(_list, &list, sizeof(list))) - err = -EFAULT; return err; } +static int snd_ctl_elem_list_user(struct snd_card *card, + struct snd_ctl_elem_list __user *_list) +{ + struct snd_ctl_elem_list list; + int err; + + if (copy_from_user(&list, _list, sizeof(list))) + return -EFAULT; + err = snd_ctl_elem_list(card, &list); + if (err) + return err; + if (copy_to_user(_list, &list, sizeof(list))) + return -EFAULT; + + return 0; +} + /* Check whether the given kctl info is valid */ static int snd_ctl_check_elem_info(struct snd_card *card, const struct snd_ctl_elem_info *info) @@ -1703,7 +1715,7 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg case SNDRV_CTL_IOCTL_CARD_INFO: return snd_ctl_card_info(card, ctl, cmd, argp); case SNDRV_CTL_IOCTL_ELEM_LIST: - return snd_ctl_elem_list(card, argp); + return snd_ctl_elem_list_user(card, argp); case SNDRV_CTL_IOCTL_ELEM_INFO: return snd_ctl_elem_info_user(ctl, argp); case SNDRV_CTL_IOCTL_ELEM_READ: diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 02df1d7db9a1..1d708aab9c98 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -22,24 +22,22 @@ struct snd_ctl_elem_list32 { static int snd_ctl_elem_list_compat(struct snd_card *card, struct snd_ctl_elem_list32 __user *data32) { - struct snd_ctl_elem_list __user *data; + struct snd_ctl_elem_list data = {}; compat_caddr_t ptr; int err; - data = compat_alloc_user_space(sizeof(*data)); - /* offset, space, used, count */ - if (copy_in_user(data, data32, 4 * sizeof(u32))) + if (copy_from_user(&data, data32, 4 * sizeof(u32))) return -EFAULT; /* pids */ - if (get_user(ptr, &data32->pids) || - put_user(compat_ptr(ptr), &data->pids)) + if (get_user(ptr, &data32->pids)) return -EFAULT; - err = snd_ctl_elem_list(card, data); + data.pids = compat_ptr(ptr); + err = snd_ctl_elem_list(card, &data); if (err < 0) return err; /* copy the result */ - if (copy_in_user(data32, data, 4 * sizeof(u32))) + if (copy_to_user(data32, &data, 4 * sizeof(u32))) return -EFAULT; return 0; } diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 21edb8ac95eb..0c029892880a 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -203,28 +203,35 @@ static int snd_hwdep_dsp_status(struct snd_hwdep *hw, } static int snd_hwdep_dsp_load(struct snd_hwdep *hw, - struct snd_hwdep_dsp_image __user *_info) + struct snd_hwdep_dsp_image *info) { - struct snd_hwdep_dsp_image info; int err; if (! hw->ops.dsp_load) return -ENXIO; - memset(&info, 0, sizeof(info)); - if (copy_from_user(&info, _info, sizeof(info))) - return -EFAULT; - if (info.index >= 32) + if (info->index >= 32) return -EINVAL; /* check whether the dsp was already loaded */ - if (hw->dsp_loaded & (1u << info.index)) + if (hw->dsp_loaded & (1u << info->index)) return -EBUSY; - err = hw->ops.dsp_load(hw, &info); + err = hw->ops.dsp_load(hw, info); if (err < 0) return err; - hw->dsp_loaded |= (1u << info.index); + hw->dsp_loaded |= (1u << info->index); return 0; } +static int snd_hwdep_dsp_load_user(struct snd_hwdep *hw, + struct snd_hwdep_dsp_image __user *_info) +{ + struct snd_hwdep_dsp_image info = {}; + + if (copy_from_user(&info, _info, sizeof(info))) + return -EFAULT; + return snd_hwdep_dsp_load(hw, &info); +} + + static long snd_hwdep_ioctl(struct file * file, unsigned int cmd, unsigned long arg) { @@ -238,7 +245,7 @@ static long snd_hwdep_ioctl(struct file * file, unsigned int cmd, case SNDRV_HWDEP_IOCTL_DSP_STATUS: return snd_hwdep_dsp_status(hw, argp); case SNDRV_HWDEP_IOCTL_DSP_LOAD: - return snd_hwdep_dsp_load(hw, argp); + return snd_hwdep_dsp_load_user(hw, argp); } if (hw->ops.ioctl) return hw->ops.ioctl(hw, file, cmd, arg); diff --git a/sound/core/hwdep_compat.c b/sound/core/hwdep_compat.c index bc81db9cb3d4..a0b76706c083 100644 --- a/sound/core/hwdep_compat.c +++ b/sound/core/hwdep_compat.c @@ -19,26 +19,17 @@ struct snd_hwdep_dsp_image32 { static int snd_hwdep_dsp_load_compat(struct snd_hwdep *hw, struct snd_hwdep_dsp_image32 __user *src) { - struct snd_hwdep_dsp_image __user *dst; + struct snd_hwdep_dsp_image info = {}; compat_caddr_t ptr; - u32 val; - dst = compat_alloc_user_space(sizeof(*dst)); - - /* index and name */ - if (copy_in_user(dst, src, 4 + 64)) - return -EFAULT; - if (get_user(ptr, &src->image) || - put_user(compat_ptr(ptr), &dst->image)) - return -EFAULT; - if (get_user(val, &src->length) || - put_user(val, &dst->length)) - return -EFAULT; - if (get_user(val, &src->driver_data) || - put_user(val, &dst->driver_data)) + if (copy_from_user(&info, src, 4 + 64) || + get_user(ptr, &src->image) || + get_user(info.length, &src->length) || + get_user(info.driver_data, &src->driver_data)) return -EFAULT; + info.image = compat_ptr(ptr); - return snd_hwdep_dsp_load(hw, dst); + return snd_hwdep_dsp_load(hw, &info); } enum { -- 2.27.0

2 1

[PATCH] sound/pci/asihpi: fix spellint typo in comments
by Wang Qing 21 Sep '20

21 Sep '20

Change the comment typo: "ununsed" -> "unused". Signed-off-by: Wang Qing <wangqing(a)vivo.com> --- sound/pci/asihpi/hpios.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index 26f7cf4..9e551bc --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -67,7 +67,7 @@ struct hpi_ioctl_linux { }; /* Conflict?: H is already used by a number of drivers hid, bluetooth hci, - and some sound drivers sb16, hdsp, emu10k. AFAIK 0xFC is ununsed command + and some sound drivers sb16, hdsp, emu10k. AFAIK 0xFC is unused command */ #define HPI_IOCTL_LINUX _IOWR('H', 0xFC, struct hpi_ioctl_linux) -- 2.7.4

2 1

[PATCH] ALSA: hda/realtek: Enable front panel headset LED on Lenovo ThinkStation P520
by Kai-Heng Feng 21 Sep '20

21 Sep '20

On Lenovo P520, the front panel headset LED isn't lit up right now. Realtek states that the LED needs to be enabled by ALC233's GPIO2, so let's do it accordingly to light the LED up. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c521a1f17096..ba941bd0b792 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6017,6 +6017,7 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, #include "hp_x360_helper.c" enum { + ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, ALC269_FIXUP_DELL_M101Z, @@ -6194,6 +6195,10 @@ enum { }; static const struct hda_fixup alc269_fixups[] = { + [ALC269_FIXUP_GPIO2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio2, + }, [ALC269_FIXUP_SONY_VAIO] = { .type = HDA_FIXUP_PINCTLS, .v.pins = (const struct hda_pintbl[]) { @@ -7013,6 +7018,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC233_FIXUP_LENOVO_MULTI_CODECS] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_alc662_fixup_lenovo_dual_codecs, + .chained = true, + .chain_id = ALC269_FIXUP_GPIO2 }, [ALC233_FIXUP_ACER_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, -- 2.17.1

3 5

[PATCH] ALSA: hda/realtek - Couldn't detect Mic if booting with headset plugged
by Hui Wang 21 Sep '20

21 Sep '20

We found a Mic detection issue on many Lenovo laptops, those laptops belong to differnt models and they have different audio design like internal mic connects to the codec or PCH, they all have this problem, the problem is if plugging a headset before powerup/reboot the machine, after booting up, the headphone could be detected but Mic couldn't. If we plug out and plug in the headset, both headphone and Mic could be detected then. Through debugging we found the codec on those laptops are same, it is alc257, and if we don't disable the 3k pulldown in alc256_shutup(), the issue will be fixed. So far there is no pop noise or power consumption regression on those laptops after this change. Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 85e207173f5d..b6dc47da1d7b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3428,7 +3428,11 @@ static void alc256_shutup(struct hda_codec *codec) /* 3k pull low control for Headset jack. */ /* NOTE: call this before clearing the pin, otherwise codec stalls */ - alc_update_coef_idx(codec, 0x46, 0, 3 << 12); + /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly + * when booting with headset plugged. So skip setting it for the codec alc257 + */ + if (codec->core.vendor_id != 0x10ec0257) + alc_update_coef_idx(codec, 0x46, 0, 3 << 12); if (!spec->no_shutup_pins) snd_hda_codec_write(codec, hp_pin, 0, -- 2.17.1

3 2

[PATCH] ALSA: asihpi: fix iounmap in error handler
by trix＠redhat.com 21 Sep '20

21 Sep '20

From: Tom Rix <trix(a)redhat.com> clang static analysis flags this problem hpioctl.c:513:7: warning: Branch condition evaluates to a garbage value if (pci.ap_mem_base[idx]) { ^~~~~~~~~~~~~~~~~~~~ If there is a failure in the middle of the memory space loop, only some of the memory spaces need to be cleaned up. At the error handler, idx holds the number of successful memory spaces mapped. So rework the handler loop to use the old idx. There is a second problem, the memory space loop conditionally iomaps()/sets the mem_base so it is necessay to initize pci. Fixes: 719f82d3987a ("ALSA: Add support of AudioScience ASI boards") Signed-off-by: Tom Rix <trix(a)redhat.com> --- sound/pci/asihpi/hpioctl.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 6cc2b6964bb5..bb31b7fe867d 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -352,7 +352,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, struct hpi_message hm; struct hpi_response hr; struct hpi_adapter adapter; - struct hpi_pci pci; + struct hpi_pci pci = { 0 }; memset(&adapter, 0, sizeof(adapter)); @@ -509,7 +509,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, return 0; err: - for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { + while (--idx >= 0) { if (pci.ap_mem_base[idx]) { iounmap(pci.ap_mem_base[idx]); pci.ap_mem_base[idx] = NULL; -- 2.18.1

2 1

[PATCH v3 00/19] Add Allwinner H3/H5/H6/A64 HDMI audio
by Clément Péron 20 Sep '20

20 Sep '20

Hi, New test done by Maxime using TDM show that's LRCK is indeed inverted so I drop the patch reverted in v2. And HDMI requires an inverted LRCK so let's readd the frame-inversion in the device-tree. I have also added a patch to change set_chan_cfg. Please note that I can't test TDM and only have a Allwinner H6. So test and comment on other Allwinner chips are welcome! Regards, Clement Change since v2: - rebase on next-20200918 - drop revert LRCK polarity patch - readd simple-audio-card,frame-inversion in dts - Add patch for changing set_chan_cfg params Change since v1: - rebase on next-20200828 - add revert LRCK polarity - remove all simple-audio-card,frame-inversion in dts - add Ondrej patches for Orange Pi board - Add arm64 defconfig patch Clément Péron (4): ASoC: sun4i-i2s: Change set_chan_cfg params ASoC: sun4i-i2s: Fix sun8i volatile regs arm64: dts: allwinner: h6: Enable HDMI sound for Beelink GS1 arm64: defconfig: Enable Allwinner i2s driver Jernej Skrabec (3): ASoC: sun4i-i2s: Add support for H6 I2S dt-bindings: ASoC: sun4i-i2s: Add H6 compatible arm64: dts: allwinner: h6: Add HDMI audio node Marcus Cooper (9): ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support arm: dts: sunxi: h3/h5: Add DAI node for HDMI arm: dts: sunxi: h3/h5: Add HDMI audio arm64: dts: allwinner: a64: Add DAI node for HDMI arm64: dts: allwinner: a64: Add HDMI audio arm: sun8i: h3: Add HDMI audio to Orange Pi 2 arm: sun8i: h3: Add HDMI audio to Beelink X2 arm64: dts: allwinner: a64: Add HDMI audio to Pine64 Ondrej Jirman (3): arm64: dts: allwinner: Enable HDMI audio on Orange Pi PC 2 ARM: dts: sun8i-h3: Enable HDMI audio on Orange Pi PC/One arm64: dts: sun50i-h6-orangepi-3: Enable HDMI audio .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm/boot/dts/sun8i-h3-beelink-x2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-2.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-one.dts | 8 + arch/arm/boot/dts/sun8i-h3-orangepi-pc.dts | 8 + arch/arm/boot/dts/sunxi-h3-h5.dtsi | 33 +++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 34 +++ .../dts/allwinner/sun50i-h5-orangepi-pc2.dts | 8 + .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 8 + .../dts/allwinner/sun50i-h6-orangepi-3.dts | 8 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 33 +++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-i2s.c | 280 ++++++++++++++++-- 14 files changed, 427 insertions(+), 20 deletions(-) -- 2.25.1

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Re: [PATCH v3 3/7] ASoC: sun4i-i2s: Add support for H6 I2S
by Samuel Holland 20 Sep '20

20 Sep '20

Maxime, On 9/10/20 9:33 AM, Maxime Ripard wrote: > On Thu, Sep 03, 2020 at 09:54:39PM -0500, Samuel Holland wrote: >> On 9/3/20 3:58 PM, Maxime Ripard wrote: >>> On Thu, Sep 03, 2020 at 10:02:31PM +0200, Clément Péron wrote: >>>> Hi Maxime, >>>> >>>> On Wed, 29 Jul 2020 at 17:16, Mark Brown <broonie(a)kernel.org> wrote: >>>>> >>>>> On Wed, Jul 29, 2020 at 04:39:27PM +0200, Maxime Ripard wrote: >>>>> >>>>>> It really looks like the polarity of LRCK is fine though. The first word >>>>>> is sent with LRCK low, and then high, so we have channel 0 and then >>>>>> channel 1 which seems to be the proper ordering? >> >> Which image file is this in reference to? >> >>>>> Yes, that's normal. >>>> >>>> Thank you very much for this test. >>>> >>>> So I will revert the following commit: >>>> >>>> ASoC: sun4i-i2s: Fix the LRCK polarity >>>> >>>> https://github.com/clementperon/linux/commit/dd657eae8164f7e4bafe8b875031a7… >>> >>> Like I said, the current code is working as expected with regard to the >>> LRCK polarity. The issue is that the samples are delayed and start to be >>> transmitted on the wrong phase of the signal. >> >> Since an I2S LRCK frame is radially symmetric, "wrong phase" and "inverted >> polarity" look the same. The only way to definitively distinguish them is by >> looking at the sample data. >> >> In "i2s-h6.png", the samples are all zeroes, so you're assuming that the first >> sample transmitted (that is, when the bit clock starts transitioning) was a >> "left" sample. >> >> However, in "h6-i2s-start-data.png", there are pairs of samples we can look at. >> I'm still assuming that similar samples are a left/right pair, but that's >> probably a safe assumption. Here we see the first sample in each pair is >> transmitted with LRCK *high*, and the second sample in the pair is transmitted >> with LRCK *low*. This is the opposite of your claim above. >> >> An ideal test would put left/right markers and frame numbers in the data >> channel. The Python script below can generate such a file. Then you would know >> how much startup delay there is, which channel the "first sample" came from, and >> how each channel maps to the LRCK level. >> >> It would also be helpful to test DSP_A mode, where the LRCK signal is >> asymmetric and an inversion would be obvious. > > I had no idea that there was a wave module in Python, that's a great > suggestion, thanks! > > You'll find attached the screenshots for both the I2S and DSP_A formats. > I zoomed out a bit to be able to have the first valid samples, but it > should be readable. > > The code I used is there: > https://github.com/mripard/linux/tree/sunxi/h6-i2s-test > > It's basically the v3, plus the DT bits. > > As you can see, in the i2s case, LRCK starts low and then goes up, with > the first channel (0x2*** samples) transmitted first, so everything > looks right here. > > On the DSP_A screenshot, LRCK will be low with small bursts high, and > once again with the first channel being transmitted first, so it looks > right to me too. Indeed, for H6 i2s0 with LRCK inversion in software, everything looks correct on the wire. It's still concerning to me that the BSP has no evidence of this inversion, either for i2s0 or i2s1[1]. And the inversion seems not to be required for HDMI audio on mainline either (but there could be an inversion on the HDMI side or on the interconnect). Even so, your research is sufficient justification for me that the code is correct as-is (with the inversion). Thank you very much for collecting the data! Cheers, Samuel [1]: https://github.com/Allwinner-Homlet/H6-BSP4.9-linux/blob/e634a960316dddd1eb… where 1 == SND_SOC_DAIFMT_NB_NF, and there's no inversion in sunxi_daudio_init_fmt().

3 4

USB-Audio.conf patch for ASUS Xonar SE IEC958
by Brian McCarter 20 Sep '20

20 Sep '20

Hello, I recently purchased an ASUS Xonar SE sound card and found that the IEC958 output did not work under ALSA. The attached patch to alsa-lib enables the output. I can see that the SE is among several other Xonar models which require special treatment to enable the S/PDIF output, but this one appears to be the only one which puts the digital output on PCM #2. Brian McCarter

1 0

[PATCH 1/2] ASoC: tas2770: Fix calling reset in probe
by Dan Murphy 18 Sep '20

18 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

1 2

[PATCH 0/2] DAI driver for new XCVR IP
by Viorel Suman (OSS) 18 Sep '20

18 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 .../devicetree/bindings/sound/fsl,xcvr.yaml | 104 ++ sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_xcvr.c | 1352 ++++++++++++++++++++ sound/soc/fsl/fsl_xcvr.h | 266 ++++ 5 files changed, 1734 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

5 10

[PATCH 0/7] SOF fixes and updates
by Ranjani Sridharan 18 Sep '20

18 Sep '20

This series includes fixes for error reporting, topology parsing and runtime PM issues along with updates for DMIC support and IMX platforms. Iulian Olaru (2): ASoC: SOF: imx: Replace sdev->private with sdev->pdata->hw_pdata ASoC: SOF: sof-of-dev: Add .arch_ops field Jaska Uimonen (1): ASoC: SOF: intel: hda: support also devices with 1 and 3 dmics Keyon Jie (1): ASoC: SOF: topology: fix the ipc_size calculation for process component Rander Wang (1): ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work Ranjani Sridharan (2): ASoC: SOF: Intel: hda: report error only for the last ROM init iteration ASoC: SOF: Intel: hda: add extended rom status dump to error log sound/soc/sof/imx/Kconfig | 2 ++ sound/soc/sof/imx/imx8.c | 17 +++++++++---- sound/soc/sof/imx/imx8m.c | 10 +++++--- sound/soc/sof/intel/hda-codec.c | 4 +-- sound/soc/sof/intel/hda-loader.c | 42 +++++++++++++++++++------------- sound/soc/sof/intel/hda.c | 26 +++++++++++++++++++- sound/soc/sof/topology.c | 4 +-- 7 files changed, 74 insertions(+), 31 deletions(-) -- 2.25.1

3 10

[PATCH v2] ALSA: HDA: Early Forbid of runtime PM
by Harsha Priya 18 Sep '20

18 Sep '20

For certain codecs (like Realtek), pm_runtime_forbid() is invoked in the probe function after build_controls(). In a stress test, its observed occasionally that runtime PM calls are invoked before controls are built. This causes the codec to be runtime suspended before probe completes. Because of this, not all controls are enumerated correctly, and audio does not work until system is rebooted. This issue being common across all codecs, pm_runtime_forbid() is called when the codec object is created to fix this issue. A codec enables or disables runtime pm in its own probe function. Multiple stress tests of 2000+ cycles has been done to test the fix. Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Emmanuel Jillela <emmanuel.jillela(a)intel.com> Reviewed-by: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/hda_codec.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index e96a87f1b611..a356c21edb90 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1000,6 +1000,9 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (err < 0) goto error; + /* PM runtime needs to be enabled later after binding codec */ + pm_runtime_forbid(&codec->core.dev); + return 0; error: -- 2.17.1

3 3

[PATCH] ASoC: fix kconfig dependency warnings for SND_SOC_WM8731
by Necip Fazil Yildiran 18 Sep '20

18 Sep '20

SND_SOC_WM8731 was made visible and dependent on other symbols with commit 1291e14175e6 ("ASoC: codecs: Make OF supported CODECs visible in Kconfig"). To this respect, the symbols selecting SND_SOC_WM8731 turned out to be overlooking its dependencies. For example, enabling SND_SOC_DB1200 and disabling SND_SOC_I2C_AND_SPI results in the following Kconfig warning since SND_SOC_DB1200 selects SND_SOC_WM8731 but does not account for its dependencies: WARNING: unmet direct dependencies detected for SND_SOC_WM8731 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_I2C_AND_SPI [=n] Selected by [y]: - SND_SOC_DB1200 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && SND_SOC_AU1XPSC [=y] Switch reverse dependencies on SND_SOC_WM8731 to normal dependencies. Fixes: 1291e14175e6 ("ASoC: codecs: Make OF supported CODECs visible in Kconfig") Signed-off-by: Necip Fazil Yildiran <fazilyildiran(a)gmail.com> --- sound/soc/atmel/Kconfig | 7 +++---- sound/soc/au1x/Kconfig | 3 +-- sound/soc/pxa/Kconfig | 4 ++-- 3 files changed, 6 insertions(+), 8 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 71f2d42188c4..abb1284835c0 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -44,8 +44,8 @@ config SND_AT91_SOC_SAM9G20_WM8731 tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board" depends on ARCH_AT91 || COMPILE_TEST depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI + depends on SND_SOC_WM8731 select SND_ATMEL_SOC_SSC_PDC - select SND_SOC_WM8731 help Say Y if you want to add support for SoC audio on WM8731-based AT91sam9g20 evaluation board. @@ -64,8 +64,8 @@ config SND_AT91_SOC_SAM9X5_WM8731 tristate "SoC Audio support for WM8731-based at91sam9x5 board" depends on ARCH_AT91 || COMPILE_TEST depends on ATMEL_SSC && SND_SOC_I2C_AND_SPI + depends on SND_SOC_WM8731 select SND_ATMEL_SOC_SSC_DMA - select SND_SOC_WM8731 help Say Y if you want to add support for audio SoC on an at91sam9x5 based board that is using WM8731 codec. @@ -110,8 +110,7 @@ config SND_ATMEL_SOC_I2S config SND_SOC_MIKROE_PROTO tristate "Support for Mikroe-PROTO board" depends on OF - depends on SND_SOC_I2C_AND_SPI - select SND_SOC_WM8731 + depends on SND_SOC_I2C_AND_SPI && SND_SOC_WM8731 help Say Y or M if you want to add support for MikroElektronika PROTO Audio Board. This board contains the WM8731 codec, which can be configured diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 38de7c0efbc7..749e59fddfcd 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -53,12 +53,11 @@ config SND_SOC_DB1000 config SND_SOC_DB1200 tristate "DB1200/DB1300/DB1550 Audio support" - depends on SND_SOC_AU1XPSC + depends on SND_SOC_AU1XPSC && SND_SOC_WM8731 select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC select SND_SOC_WM9712 select SND_SOC_AU1XPSC_I2S - select SND_SOC_WM8731 help Select this option to enable audio (AC97 and I2S) on the Alchemy/AMD/RMI/NetLogic Db1200, Db1550 and Db1300 evaluation boards. diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 0ac85eada75c..eeb1dde59468 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -44,8 +44,8 @@ config SND_MMP_SOC_SSPA config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx && I2C + depends on SND_SOC_WM8731 select SND_PXA2XX_SOC_I2S - select SND_SOC_WM8731 help Say Y if you want to add support for SoC audio on Sharp Zaurus SL-C7x0 models (Corgi, Shepherd, Husky). @@ -70,8 +70,8 @@ config SND_PXA2XX_SOC_Z2 config SND_PXA2XX_SOC_POODLE tristate "SoC Audio support for Poodle" depends on SND_PXA2XX_SOC && MACH_POODLE && I2C + depends on SND_SOC_WM8731 select SND_PXA2XX_SOC_I2S - select SND_SOC_WM8731 help Say Y if you want to add support for SoC audio on Sharp Zaurus SL-5600 model (Poodle). -- 2.25.1

2 1

Re: [PATCH v2 0/2] soundwire: use {u32|u16}p_replace_bits
by Liao, Bard 18 Sep '20

18 Sep '20

On 9/17/2020 8:01 PM, Vinod Koul wrote: > This is v2 of the series to fix usage of FIELD_PREP and use > {u32|u16}p_replace_bits. > > v1 used u32_replace_bits but looks like it might be removed so use the > {u32|u16}p_replace_bits() instead. Also this helped to find that one > instance in intel driver should have used u16p_replace_bits() so that is > fixed too > > Vinod Koul (2): > soundwire: cadence: use u32p_replace_bits > soundwire: intel: use {u32|u16}p_replace_bits > > drivers/soundwire/cadence_master.c | 21 ++++++++++----------- > drivers/soundwire/intel.c | 14 +++++++------- > 2 files changed, 17 insertions(+), 18 deletions(-) Tested it on a Intel platform. So Tested-by: Bard Liao <yung-chuan.liao(a)linux.intel.com>

2 1

[PATCH v4 0/3] soundwire: qcom: fix IP version v1.5.1 support
by Srinivas Kandagatla 18 Sep '20

18 Sep '20

While testing Qualcomm soundwire controller version 1.5.1, found two issue, Firstly the frame shape information configured vs the bus parameters are out of sync. secondly some ports on this ip version require block packing mode support. With this patchset I was able to test 2 WSA speakers! Also I found a regression due to move to REG_FIELD, which patch 1 fixes it! thanks, srini Change since v3: - use u32p_replace_bits instead of u32_replace_bits Srinivas Kandagatla (3): soundwire: qcom: clear BIT FIELDs before value set. soundwire: qcom: add support to block packing mode soundwire: qcom: get max rows and cols info from compatible drivers/soundwire/qcom.c | 70 +++++++++++++++++++++++++++++----------- 1 file changed, 51 insertions(+), 19 deletions(-) -- 2.21.0

2 4

[PATCH] soundwire: Add generic bandwidth allocation algorithm
by Bard Liao 18 Sep '20

18 Sep '20

This algorithm computes bus parameters like clock frequency, frame shape and port transport parameters based on active stream(s) running on the bus. Developers can also implement their own .compute_params() callback for specific resource management algorithm, and set if before calling sdw_add_bus_master() Credits: this patch is based on an earlier internal contribution by Vinod Koul, Sanyog Kale, Shreyas Nc and Hardik Shah. All hard-coded values were removed from the initial contribution to use BIOS information instead. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/Kconfig | 5 + drivers/soundwire/Makefile | 3 + drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 46 +- .../soundwire/generic_bandwidth_allocation.c | 427 ++++++++++++++++++ drivers/soundwire/intel.c | 3 + drivers/soundwire/stream.c | 12 + include/linux/soundwire/sdw.h | 3 + 8 files changed, 503 insertions(+), 2 deletions(-) create mode 100644 drivers/soundwire/generic_bandwidth_allocation.c diff --git a/drivers/soundwire/Kconfig b/drivers/soundwire/Kconfig index f83d02c9c60a..016e74230bb7 100644 --- a/drivers/soundwire/Kconfig +++ b/drivers/soundwire/Kconfig @@ -24,6 +24,7 @@ config SOUNDWIRE_CADENCE config SOUNDWIRE_INTEL tristate "Intel SoundWire Master driver" select SOUNDWIRE_CADENCE + select SOUNDWIRE_GENERIC_ALLOCATION depends on ACPI && SND_SOC help SoundWire Intel Master driver. @@ -40,4 +41,8 @@ config SOUNDWIRE_QCOM If you have an Qualcomm platform which has a SoundWire Master then enable this config option to get the SoundWire support for that device + +config SOUNDWIRE_GENERIC_ALLOCATION + tristate + endif diff --git a/drivers/soundwire/Makefile b/drivers/soundwire/Makefile index 7c53ffae9f50..bf1e250d50dd 100644 --- a/drivers/soundwire/Makefile +++ b/drivers/soundwire/Makefile @@ -8,6 +8,9 @@ soundwire-bus-y := bus_type.o bus.o master.o slave.o mipi_disco.o stream.o \ sysfs_slave.o sysfs_slave_dpn.o obj-$(CONFIG_SOUNDWIRE) += soundwire-bus.o +soundwire-generic-allocation-objs := generic_bandwidth_allocation.o +obj-$(CONFIG_SOUNDWIRE_GENERIC_ALLOCATION) += soundwire-generic-allocation.o + ifdef CONFIG_DEBUG_FS soundwire-bus-y += debugfs.o endif diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index d808f0256ba0..11c5a402afcd 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -61,6 +61,12 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, return -EINVAL; } + if (!bus->compute_params) { + dev_err(bus->dev, + "Bandwidth allocation not configured, compute_params no set\n"); + return -EINVAL; + } + mutex_init(&bus->msg_lock); mutex_init(&bus->bus_lock); INIT_LIST_HEAD(&bus->slaves); diff --git a/drivers/soundwire/bus.h b/drivers/soundwire/bus.h index 82484f741168..d2fda8c69f0d 100644 --- a/drivers/soundwire/bus.h +++ b/drivers/soundwire/bus.h @@ -69,6 +69,7 @@ struct sdw_msg { }; #define SDW_DOUBLE_RATE_FACTOR 2 +#define SDW_STRM_RATE_GROUPING 1 extern int sdw_rows[SDW_FRAME_ROWS]; extern int sdw_cols[SDW_FRAME_COLS]; @@ -154,9 +155,50 @@ int sdw_transfer_defer(struct sdw_bus *bus, struct sdw_msg *msg, int sdw_fill_msg(struct sdw_msg *msg, struct sdw_slave *slave, u32 addr, size_t count, u16 dev_num, u8 flags, u8 *buf); +/* Retrieve and return channel count from channel mask */ +static inline int sdw_ch_mask_to_ch(int ch_mask) +{ + int c = 0; + + for (c = 0; ch_mask; ch_mask >>= 1) + c += ch_mask & 1; + + return c; +} + +/* Fill transport parameter data structure */ +static inline void sdw_fill_xport_params(struct sdw_transport_params *params, + int port_num, bool grp_ctrl_valid, + int grp_ctrl, int sample_int, + int off1, int off2, + int hstart, int hstop, + int pack_mode, int lane_ctrl) +{ + params->port_num = port_num; + params->blk_grp_ctrl_valid = grp_ctrl_valid; + params->blk_grp_ctrl = grp_ctrl; + params->sample_interval = sample_int; + params->offset1 = off1; + params->offset2 = off2; + params->hstart = hstart; + params->hstop = hstop; + params->blk_pkg_mode = pack_mode; + params->lane_ctrl = lane_ctrl; +} + +/* Fill port parameter data structure */ +static inline void sdw_fill_port_params(struct sdw_port_params *params, + int port_num, int bps, + int flow_mode, int data_mode) +{ + params->num = port_num; + params->bps = bps; + params->flow_mode = flow_mode; + params->data_mode = data_mode; +} + /* Read-Modify-Write Slave register */ -static inline int -sdw_update(struct sdw_slave *slave, u32 addr, u8 mask, u8 val) +static inline int sdw_update(struct sdw_slave *slave, u32 addr, u8 mask, u8 val) { int tmp; diff --git a/drivers/soundwire/generic_bandwidth_allocation.c b/drivers/soundwire/generic_bandwidth_allocation.c new file mode 100644 index 000000000000..6088775b67a5 --- /dev/null +++ b/drivers/soundwire/generic_bandwidth_allocation.c @@ -0,0 +1,427 @@ +// SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) +// Copyright(c) 2015-2020 Intel Corporation. + +/* + * Bandwidth management algorithm based on 2^n gears + * + */ + +#include <linux/device.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/slab.h> +#include <linux/soundwire/sdw.h> +#include "bus.h" + +#define SDW_STRM_RATE_GROUPING 1 + +struct sdw_group_params { + unsigned int rate; + int full_bw; + int payload_bw; + int hwidth; +}; + +struct sdw_group { + unsigned int count; + unsigned int max_size; + unsigned int *rates; +}; + +struct sdw_transport_data { + int hstart; + int hstop; + int block_offset; + int sub_block_offset; +}; + +static void sdw_compute_slave_ports(struct sdw_master_runtime *m_rt, + struct sdw_transport_data *t_data) +{ + struct sdw_slave_runtime *s_rt = NULL; + struct sdw_port_runtime *p_rt; + int port_bo, sample_int; + unsigned int rate, bps, ch = 0; + unsigned int slave_total_ch; + + port_bo = t_data->block_offset; + + list_for_each_entry(s_rt, &m_rt->slave_rt_list, m_rt_node) { + rate = m_rt->stream->params.rate; + bps = m_rt->stream->params.bps; + sample_int = (m_rt->bus->params.curr_dr_freq / rate); + slave_total_ch = 0; + + list_for_each_entry(p_rt, &s_rt->port_list, port_node) { + ch = sdw_ch_mask_to_ch(p_rt->ch_mask); + + sdw_fill_xport_params(&p_rt->transport_params, + p_rt->num, false, + SDW_BLK_GRP_CNT_1, + sample_int, port_bo, port_bo >> 8, + t_data->hstart, + t_data->hstop, + (SDW_BLK_GRP_CNT_1 * ch), 0x0); + + sdw_fill_port_params(&p_rt->port_params, + p_rt->num, bps, + SDW_PORT_FLOW_MODE_ISOCH, + SDW_PORT_DATA_MODE_NORMAL); + + port_bo += bps * ch; + slave_total_ch += ch; + } + + if (m_rt->direction == SDW_DATA_DIR_TX && + m_rt->ch_count == slave_total_ch) { + /* + * Slave devices were configured to access all channels + * of the stream, which indicates that they operate in + * 'mirror mode'. Make sure we reset the port offset for + * the next device in the list + */ + port_bo = t_data->block_offset; + } + } +} + +static void sdw_compute_master_ports(struct sdw_master_runtime *m_rt, + struct sdw_group_params *params, + int port_bo, int hstop) +{ + struct sdw_transport_data t_data = {0}; + struct sdw_port_runtime *p_rt; + struct sdw_bus *bus = m_rt->bus; + int sample_int, hstart = 0; + unsigned int rate, bps, ch, no_ch; + + rate = m_rt->stream->params.rate; + bps = m_rt->stream->params.bps; + ch = m_rt->ch_count; + sample_int = (bus->params.curr_dr_freq / rate); + + if (rate != params->rate) + return; + + t_data.hstop = hstop; + hstart = hstop - params->hwidth + 1; + t_data.hstart = hstart; + + list_for_each_entry(p_rt, &m_rt->port_list, port_node) { + no_ch = sdw_ch_mask_to_ch(p_rt->ch_mask); + + sdw_fill_xport_params(&p_rt->transport_params, p_rt->num, + false, SDW_BLK_GRP_CNT_1, sample_int, + port_bo, port_bo >> 8, hstart, hstop, + (SDW_BLK_GRP_CNT_1 * no_ch), 0x0); + + sdw_fill_port_params(&p_rt->port_params, + p_rt->num, bps, + SDW_PORT_FLOW_MODE_ISOCH, + SDW_PORT_DATA_MODE_NORMAL); + + /* Check for first entry */ + if (!(p_rt == list_first_entry(&m_rt->port_list, + struct sdw_port_runtime, + port_node))) { + port_bo += bps * ch; + continue; + } + + t_data.hstart = hstart; + t_data.hstop = hstop; + t_data.block_offset = port_bo; + t_data.sub_block_offset = 0; + port_bo += bps * ch; + } + + sdw_compute_slave_ports(m_rt, &t_data); +} + +static void _sdw_compute_port_params(struct sdw_bus *bus, + struct sdw_group_params *params, int count) +{ + struct sdw_master_runtime *m_rt = NULL; + int hstop = bus->params.col - 1; + int block_offset, port_bo, i; + + /* Run loop for all groups to compute transport parameters */ + for (i = 0; i < count; i++) { + port_bo = 1; + block_offset = 1; + + list_for_each_entry(m_rt, &bus->m_rt_list, bus_node) { + sdw_compute_master_ports(m_rt, &params[i], + port_bo, hstop); + + block_offset += m_rt->ch_count * + m_rt->stream->params.bps; + port_bo = block_offset; + } + + hstop = hstop - params[i].hwidth; + } +} + +static int sdw_compute_group_params(struct sdw_bus *bus, + struct sdw_group_params *params, + int *rates, int count) +{ + struct sdw_master_runtime *m_rt = NULL; + int sel_col = bus->params.col; + unsigned int rate, bps, ch; + int i, column_needed = 0; + + /* Calculate bandwidth per group */ + for (i = 0; i < count; i++) { + params[i].rate = rates[i]; + params[i].full_bw = bus->params.curr_dr_freq / params[i].rate; + } + + list_for_each_entry(m_rt, &bus->m_rt_list, bus_node) { + rate = m_rt->stream->params.rate; + bps = m_rt->stream->params.bps; + ch = m_rt->ch_count; + + for (i = 0; i < count; i++) { + if (rate == params[i].rate) + params[i].payload_bw += bps * ch; + } + } + + for (i = 0; i < count; i++) { + params[i].hwidth = (sel_col * + params[i].payload_bw + params[i].full_bw - 1) / + params[i].full_bw; + + column_needed += params[i].hwidth; + } + + if (column_needed > sel_col - 1) + return -EINVAL; + + return 0; +} + +static int sdw_add_element_group_count(struct sdw_group *group, + unsigned int rate) +{ + int num = group->count; + int i; + + for (i = 0; i <= num; i++) { + if (rate == group->rates[i]) + break; + + if (i != num) + continue; + + if (group->count >= group->max_size) { + unsigned int *rates; + + group->max_size += 1; + rates = krealloc(group->rates, + (sizeof(int) * group->max_size), + GFP_KERNEL); + if (!rates) + return -ENOMEM; + group->rates = rates; + } + + group->rates[group->count++] = rate; + } + + return 0; +} + +static int sdw_get_group_count(struct sdw_bus *bus, + struct sdw_group *group) +{ + struct sdw_master_runtime *m_rt; + unsigned int rate; + int ret = 0; + + group->count = 0; + group->max_size = SDW_STRM_RATE_GROUPING; + group->rates = kcalloc(group->max_size, sizeof(int), GFP_KERNEL); + if (!group->rates) + return -ENOMEM; + + list_for_each_entry(m_rt, &bus->m_rt_list, bus_node) { + rate = m_rt->stream->params.rate; + if (m_rt == list_first_entry(&bus->m_rt_list, + struct sdw_master_runtime, + bus_node)) { + group->rates[group->count++] = rate; + + } else { + ret = sdw_add_element_group_count(group, rate); + if (ret < 0) { + kfree(group->rates); + return ret; + } + } + } + + return ret; +} + +/** + * sdw_compute_port_params: Compute transport and port parameters + * + * @bus: SDW Bus instance + */ +static int sdw_compute_port_params(struct sdw_bus *bus) +{ + struct sdw_group_params *params = NULL; + struct sdw_group group; + int ret; + + ret = sdw_get_group_count(bus, &group); + if (ret < 0) + return ret; + + if (group.count == 0) + goto out; + + params = kcalloc(group.count, sizeof(*params), GFP_KERNEL); + if (!params) { + ret = -ENOMEM; + goto out; + } + + /* Compute transport parameters for grouped streams */ + ret = sdw_compute_group_params(bus, params, + &group.rates[0], group.count); + if (ret < 0) + goto free_params; + + _sdw_compute_port_params(bus, params, group.count); + +free_params: + kfree(params); +out: + kfree(group.rates); + + return ret; +} + +static int sdw_select_row_col(struct sdw_bus *bus, int clk_freq) +{ + struct sdw_master_prop *prop = &bus->prop; + int frame_int, frame_freq; + int r, c; + + for (c = 0; c < SDW_FRAME_COLS; c++) { + for (r = 0; r < SDW_FRAME_ROWS; r++) { + if (sdw_rows[r] != prop->default_row || + sdw_cols[c] != prop->default_col) + continue; + + frame_int = sdw_rows[r] * sdw_cols[c]; + frame_freq = clk_freq / frame_int; + + if ((clk_freq - (frame_freq * SDW_FRAME_CTRL_BITS)) < + bus->params.bandwidth) + continue; + + bus->params.row = sdw_rows[r]; + bus->params.col = sdw_cols[c]; + return 0; + } + } + + return -EINVAL; +} + +/** + * sdw_compute_bus_params: Compute bus parameters + * + * @bus: SDW Bus instance + */ +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; + 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; + is_gear = true; + } else if (mstr_prop->num_clk_freq) { + clk_values = mstr_prop->num_clk_freq; + clk_buf = mstr_prop->clk_freq; + } else { + clk_values = 1; + clk_buf = NULL; + } + + max_dr_freq = mstr_prop->max_clk_freq * SDW_DOUBLE_RATE_FACTOR; + + for (i = 0; i < clk_values; i++) { + if (!clk_buf) + curr_dr_freq = max_dr_freq; + else + curr_dr_freq = (is_gear) ? + (max_dr_freq >> clk_buf[i]) : + clk_buf[i] * SDW_DOUBLE_RATE_FACTOR; + + if (curr_dr_freq <= bus->params.bandwidth) + continue; + + break; + + /* + * TODO: Check all the Slave(s) port(s) audio modes and find + * whether given clock rate is supported with glitchless + * transition. + */ + } + + if (i == clk_values) + return -EINVAL; + + ret = sdw_select_row_col(bus, curr_dr_freq); + if (ret < 0) + return -EINVAL; + + bus->params.curr_dr_freq = curr_dr_freq; + return 0; +} + +/** + * sdw_compute_params: Compute bus, transport and port parameters + * + * @bus: SDW Bus instance + */ +int sdw_compute_params(struct sdw_bus *bus) +{ + int ret; + + /* Computes clock frequency, frame shape and frame frequency */ + ret = sdw_compute_bus_params(bus); + if (ret < 0) { + dev_err(bus->dev, "Compute bus params failed: %d", ret); + return ret; + } + + /* Compute transport and port params */ + ret = sdw_compute_port_params(bus); + if (ret < 0) { + dev_err(bus->dev, "Compute transport params failed: %d", ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL(sdw_compute_params); + +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SoundWire Generic Bandwidth Allocation"); diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index e047910d73f5..1211d114ff59 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1318,6 +1318,9 @@ static int intel_master_probe(struct platform_device *pdev) /* set driver data, accessed by snd_soc_dai_get_drvdata() */ dev_set_drvdata(dev, cdns); + /* use generic bandwidth allocation algorithm */ + sdw->cdns.bus.compute_params = sdw_compute_params; + ret = sdw_bus_master_add(bus, dev, dev->fwnode); if (ret) { dev_err(dev, "sdw_bus_master_add fail: %d\n", ret); diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index b8b1973e3ee2..f3219d1fa63b 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -25,8 +25,10 @@ int sdw_rows[SDW_FRAME_ROWS] = {48, 50, 60, 64, 75, 80, 125, 147, 96, 100, 120, 128, 150, 160, 250, 0, 192, 200, 240, 256, 72, 144, 90, 180}; +EXPORT_SYMBOL(sdw_rows); int sdw_cols[SDW_FRAME_COLS] = {2, 4, 6, 8, 10, 12, 14, 16}; +EXPORT_SYMBOL(sdw_cols); int sdw_find_col_index(int col) { @@ -1782,6 +1784,16 @@ static int _sdw_deprepare_stream(struct sdw_stream_runtime *stream) bus->params.bandwidth -= m_rt->stream->params.rate * m_rt->ch_count * m_rt->stream->params.bps; + /* Compute params */ + if (bus->compute_params) { + ret = bus->compute_params(bus); + if (ret < 0) { + dev_err(bus->dev, "Compute params failed: %d", + ret); + return ret; + } + } + /* Program params */ ret = sdw_program_params(bus, false); if (ret < 0) { diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 1e9010c139f0..3fa1b18f7f97 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -955,6 +955,9 @@ struct sdw_stream_runtime { struct sdw_stream_runtime *sdw_alloc_stream(const char *stream_name); void sdw_release_stream(struct sdw_stream_runtime *stream); + +int sdw_compute_params(struct sdw_bus *bus); + int sdw_stream_add_master(struct sdw_bus *bus, struct sdw_stream_config *stream_config, struct sdw_port_config *port_config, -- 2.17.1

3 3

[PATCH v4] ASoC: tlv320adcx140: Add support for configuring GPIO pin
by Camel Guo 18 Sep '20

18 Sep '20

From: Camel Guo <camelg(a)axis.com> Add support to configure the GPIO pin to the specific configuration. The GPIO pin can be configured as GPO, IRQ, SDOUT2, PDMCLK, MICBASE_EN, GPI, MCLK, SDIN, PDMDIN1, PDMDIN2, PDMDIN3 or PDMDIN4 and the output drive can be configured with various configuration. Signed-off-by: Camel Guo <camelg(a)axis.com> Acked-by: Dan Murphy <dmurphy(a)ti.com> --- v4: - Rebase and fix merge conflict - Add Acked-by from Dan v3: - Add ADCX140_NUM_GPIO_CFGS avoiding using magic number - Remove unneeded check on ret in adcx140_configure_gpio sound/soc/codecs/tlv320adcx140.c | 40 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 5 ++++ 2 files changed, 45 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 28dbd7d5e149..53a80246aee1 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -861,6 +861,42 @@ static int adcx140_configure_gpo(struct adcx140_priv *adcx140) } +static int adcx140_configure_gpio(struct adcx140_priv *adcx140) +{ + int gpio_count = 0; + u32 gpio_outputs[ADCX140_NUM_GPIO_CFGS]; + u32 gpio_output_val = 0; + int ret; + + gpio_count = device_property_count_u32(adcx140->dev, + "ti,gpio-config"); + if (gpio_count == 0) + return 0; + + if (gpio_count != ADCX140_NUM_GPIO_CFGS) + return -EINVAL; + + ret = device_property_read_u32_array(adcx140->dev, "ti,gpio-config", + gpio_outputs, gpio_count); + if (ret) + return ret; + + if (gpio_outputs[0] > ADCX140_GPIO_CFG_MAX) { + dev_err(adcx140->dev, "GPIO config out of range\n"); + return -EINVAL; + } + + if (gpio_outputs[1] > ADCX140_GPIO_DRV_MAX) { + dev_err(adcx140->dev, "GPIO drive out of range\n"); + return -EINVAL; + } + + gpio_output_val = gpio_outputs[0] << ADCX140_GPIO_SHIFT + | gpio_outputs[1]; + + return regmap_write(adcx140->regmap, ADCX140_GPIO_CFG0, gpio_output_val); +} + static int adcx140_codec_probe(struct snd_soc_component *component) { struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); @@ -958,6 +994,10 @@ static int adcx140_codec_probe(struct snd_soc_component *component) return ret; } + ret = adcx140_configure_gpio(adcx140); + if (ret) + return ret; + ret = adcx140_configure_gpo(adcx140); if (ret) goto out; diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index 107bd7927d9c..d7d4e3a88b5c 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -148,4 +148,9 @@ #define ADCX140_TX_FILL BIT(0) +#define ADCX140_NUM_GPIO_CFGS 2 +#define ADCX140_GPIO_SHIFT 4 +#define ADCX140_GPIO_CFG_MAX 15 +#define ADCX140_GPIO_DRV_MAX 5 + #endif /* _TLV320ADCX140_ */ -- 2.20.1

3 3

[PATCH v4 1/2] dt-bindings: tlv320adcx140: Add GPIO config and drive config
by Camel Guo 18 Sep '20

18 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

1 0

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 18 Sep '20

18 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: I am encountering an unexpected behaviour with pulse audio master volume in my application. ### Description If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel as we can expected. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH AUTOSEL 4.4 52/64] ALSA: hda: Fix potential race in unsol event handler
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c637fa151259c0f74665fde7cba5b7eac1417ae5 ] The unsol event handling code has a loop retrieving the read/write indices and the arrays without locking while the append to the array may happen concurrently. This may lead to some inconsistency. Although there hasn't been any proof of this bad results, it's still safer to protect the racy accesses. This patch adds the spinlock protection around the unsol handling loop for addressing it. Here we take bus->reg_lock as the writer side snd_hdac_bus_queue_event() is also protected by that lock. Link: https://lore.kernel.org/r/20200516062556.30951-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_bus.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 0e81ea89a5965..e3f68a76d90eb 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -155,6 +155,7 @@ static void process_unsol_events(struct work_struct *work) struct hdac_driver *drv; unsigned int rp, caddr, res; + spin_lock_irq(&bus->reg_lock); while (bus->unsol_rp != bus->unsol_wp) { rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE; bus->unsol_rp = rp; @@ -166,10 +167,13 @@ static void process_unsol_events(struct work_struct *work) codec = bus->caddr_tbl[caddr & 0x0f]; if (!codec || !codec->dev.driver) continue; + spin_unlock_irq(&bus->reg_lock); drv = drv_to_hdac_driver(codec->dev.driver); if (drv->unsol_event) drv->unsol_event(codec, res); + spin_lock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); } /** -- 2.25.1

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[PATCH AUTOSEL 4.4 39/64] ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
by Sasha Levin 18 Sep '20

18 Sep '20

From: Andreas Steinmetz <ast(a)domdv.de> [ Upstream commit 5c6cd7021a05a02fcf37f360592d7c18d4d807fb ] The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@do… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/midi.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 5c4a3d6c42341..934540042bc2e 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1803,6 +1803,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1840,11 +1862,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { -- 2.25.1

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[PATCH AUTOSEL 4.4 04/64] ASoC: kirkwood: fix IRQ error handling
by Sasha Levin 18 Sep '20

18 Sep '20

From: Russell King <rmk+kernel(a)armlinux.org.uk> [ Upstream commit 175fc928198236037174e5c5c066fe3c4691903e ] Propagate the error code from request_irq(), rather than returning -EBUSY. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> Link: https://lore.kernel.org/r/E1iNIqh-0000tW-EZ@rmk-PC.armlinux.org.uk Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index dbfdfe99c69df..231c7d97333c7 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -136,7 +136,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but -- 2.25.1

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[PATCH AUTOSEL 4.9 74/90] ALSA: hda: Fix potential race in unsol event handler
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c637fa151259c0f74665fde7cba5b7eac1417ae5 ] The unsol event handling code has a loop retrieving the read/write indices and the arrays without locking while the append to the array may happen concurrently. This may lead to some inconsistency. Although there hasn't been any proof of this bad results, it's still safer to protect the racy accesses. This patch adds the spinlock protection around the unsol handling loop for addressing it. Here we take bus->reg_lock as the writer side snd_hdac_bus_queue_event() is also protected by that lock. Link: https://lore.kernel.org/r/20200516062556.30951-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_bus.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 0e81ea89a5965..e3f68a76d90eb 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -155,6 +155,7 @@ static void process_unsol_events(struct work_struct *work) struct hdac_driver *drv; unsigned int rp, caddr, res; + spin_lock_irq(&bus->reg_lock); while (bus->unsol_rp != bus->unsol_wp) { rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE; bus->unsol_rp = rp; @@ -166,10 +167,13 @@ static void process_unsol_events(struct work_struct *work) codec = bus->caddr_tbl[caddr & 0x0f]; if (!codec || !codec->dev.driver) continue; + spin_unlock_irq(&bus->reg_lock); drv = drv_to_hdac_driver(codec->dev.driver); if (drv->unsol_event) drv->unsol_event(codec, res); + spin_lock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); } /** -- 2.25.1

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[PATCH AUTOSEL 4.9 61/90] ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
by Sasha Levin 18 Sep '20

18 Sep '20

From: Andreas Steinmetz <ast(a)domdv.de> [ Upstream commit 5c6cd7021a05a02fcf37f360592d7c18d4d807fb ] The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@do… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/midi.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 0676e7d485def..b8d4b5b3e54a1 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1805,6 +1805,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1842,11 +1864,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { -- 2.25.1

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[PATCH AUTOSEL 4.9 32/90] ALSA: hda: Clear RIRB status before reading WP
by Sasha Levin 18 Sep '20

18 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 6d011d5057ff88ee556c000ac6fe0be23bdfcd72 ] RIRB interrupt status getting cleared after the write pointer is read causes a race condition, where last response(s) into RIRB may remain unserviced by IRQ, eventually causing azx_rirb_get_response to fall back to polling mode. Clearing the RIRB interrupt status ahead of write pointer access ensures that this condition is avoided. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Signed-off-by: Viswanath L <viswanathl(a)nvidia.com> Link: https://lore.kernel.org/r/1580983853-351-1-git-send-email-viswanathl@nvidia… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index bd0e4710d15d7..79043b481d7b6 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1158,16 +1158,23 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update)) active = true; - /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures that no + * interrupt gets masked after the RIRB wp is read in + * snd_hdac_bus_update_rirb. This avoids a possible + * race condition where codec response in RIRB may + * remain unserviced by IRQ, eventually falling back + * to polling mode in azx_rirb_get_response. + */ + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); active = true; if (status & RIRB_INT_RESPONSE) { if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) udelay(80); snd_hdac_bus_update_rirb(bus); } - azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } } while (active && ++repeat < 10); -- 2.25.1

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[PATCH AUTOSEL 4.9 04/90] ASoC: kirkwood: fix IRQ error handling
by Sasha Levin 18 Sep '20

18 Sep '20

From: Russell King <rmk+kernel(a)armlinux.org.uk> [ Upstream commit 175fc928198236037174e5c5c066fe3c4691903e ] Propagate the error code from request_irq(), rather than returning -EBUSY. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> Link: https://lore.kernel.org/r/E1iNIqh-0000tW-EZ@rmk-PC.armlinux.org.uk Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index dafd22e874e99..e655425e4819e 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -136,7 +136,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but -- 2.25.1

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[PATCH AUTOSEL 4.14 101/127] ALSA: hda: Fix potential race in unsol event handler
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c637fa151259c0f74665fde7cba5b7eac1417ae5 ] The unsol event handling code has a loop retrieving the read/write indices and the arrays without locking while the append to the array may happen concurrently. This may lead to some inconsistency. Although there hasn't been any proof of this bad results, it's still safer to protect the racy accesses. This patch adds the spinlock protection around the unsol handling loop for addressing it. Here we take bus->reg_lock as the writer side snd_hdac_bus_queue_event() is also protected by that lock. Link: https://lore.kernel.org/r/20200516062556.30951-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_bus.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 714a51721a313..ab9236e4c157e 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -155,6 +155,7 @@ static void process_unsol_events(struct work_struct *work) struct hdac_driver *drv; unsigned int rp, caddr, res; + spin_lock_irq(&bus->reg_lock); while (bus->unsol_rp != bus->unsol_wp) { rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE; bus->unsol_rp = rp; @@ -166,10 +167,13 @@ static void process_unsol_events(struct work_struct *work) codec = bus->caddr_tbl[caddr & 0x0f]; if (!codec || !codec->dev.driver) continue; + spin_unlock_irq(&bus->reg_lock); drv = drv_to_hdac_driver(codec->dev.driver); if (drv->unsol_event) drv->unsol_event(codec, res); + spin_lock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); } /** -- 2.25.1

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[PATCH AUTOSEL 4.14 079/127] ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
by Sasha Levin 18 Sep '20

18 Sep '20

From: Andreas Steinmetz <ast(a)domdv.de> [ Upstream commit 5c6cd7021a05a02fcf37f360592d7c18d4d807fb ] The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@do… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/midi.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 1bfae7a1c32f1..a3d1c0c1b4a67 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1805,6 +1805,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1842,11 +1864,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { -- 2.25.1

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[PATCH AUTOSEL 4.14 041/127] ALSA: hda: Clear RIRB status before reading WP
by Sasha Levin 18 Sep '20

18 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 6d011d5057ff88ee556c000ac6fe0be23bdfcd72 ] RIRB interrupt status getting cleared after the write pointer is read causes a race condition, where last response(s) into RIRB may remain unserviced by IRQ, eventually causing azx_rirb_get_response to fall back to polling mode. Clearing the RIRB interrupt status ahead of write pointer access ensures that this condition is avoided. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Signed-off-by: Viswanath L <viswanathl(a)nvidia.com> Link: https://lore.kernel.org/r/1580983853-351-1-git-send-email-viswanathl@nvidia… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index fa261b27d8588..8198d2e53b7df 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1169,16 +1169,23 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update)) active = true; - /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures that no + * interrupt gets masked after the RIRB wp is read in + * snd_hdac_bus_update_rirb. This avoids a possible + * race condition where codec response in RIRB may + * remain unserviced by IRQ, eventually falling back + * to polling mode in azx_rirb_get_response. + */ + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); active = true; if (status & RIRB_INT_RESPONSE) { if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) udelay(80); snd_hdac_bus_update_rirb(bus); } - azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } } while (active && ++repeat < 10); -- 2.25.1

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[PATCH AUTOSEL 4.14 005/127] ASoC: kirkwood: fix IRQ error handling
by Sasha Levin 18 Sep '20

18 Sep '20

From: Russell King <rmk+kernel(a)armlinux.org.uk> [ Upstream commit 175fc928198236037174e5c5c066fe3c4691903e ] Propagate the error code from request_irq(), rather than returning -EBUSY. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> Link: https://lore.kernel.org/r/E1iNIqh-0000tW-EZ@rmk-PC.armlinux.org.uk Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index cf23af159acf4..35ca8e8bb5e52 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -136,7 +136,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 185/206] ASoC: img-i2s-out: Fix runtime PM imbalance on error
by Sasha Levin 18 Sep '20

18 Sep '20

From: Dinghao Liu <dinghao.liu(a)zju.edu.cn> [ Upstream commit 65bd91dd6957390c42a0491b9622cf31a2cdb140 ] pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> Link: https://lore.kernel.org/r/20200529012230.5863-1-dinghao.liu@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/img/img-i2s-out.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index fc2d1dac63339..798ab579564cb 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -350,8 +350,10 @@ static int img_i2s_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) chan_control_mask = IMG_I2S_OUT_CHAN_CTL_CLKT_MASK; ret = pm_runtime_get_sync(i2s->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(i2s->dev); return ret; + } img_i2s_out_disable(i2s); @@ -491,8 +493,10 @@ static int img_i2s_out_probe(struct platform_device *pdev) goto err_pm_disable; } ret = pm_runtime_get_sync(&pdev->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(&pdev->dev); goto err_suspend; + } reg = IMG_I2S_OUT_CTL_FRM_SIZE_MASK; img_i2s_out_writel(i2s, reg, IMG_I2S_OUT_CTL); -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 164/206] ALSA: hda: Fix potential race in unsol event handler
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c637fa151259c0f74665fde7cba5b7eac1417ae5 ] The unsol event handling code has a loop retrieving the read/write indices and the arrays without locking while the append to the array may happen concurrently. This may lead to some inconsistency. Although there hasn't been any proof of this bad results, it's still safer to protect the racy accesses. This patch adds the spinlock protection around the unsol handling loop for addressing it. Here we take bus->reg_lock as the writer side snd_hdac_bus_queue_event() is also protected by that lock. Link: https://lore.kernel.org/r/20200516062556.30951-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_bus.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 714a51721a313..ab9236e4c157e 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -155,6 +155,7 @@ static void process_unsol_events(struct work_struct *work) struct hdac_driver *drv; unsigned int rp, caddr, res; + spin_lock_irq(&bus->reg_lock); while (bus->unsol_rp != bus->unsol_wp) { rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE; bus->unsol_rp = rp; @@ -166,10 +167,13 @@ static void process_unsol_events(struct work_struct *work) codec = bus->caddr_tbl[caddr & 0x0f]; if (!codec || !codec->dev.driver) continue; + spin_unlock_irq(&bus->reg_lock); drv = drv_to_hdac_driver(codec->dev.driver); if (drv->unsol_event) drv->unsol_event(codec, res); + spin_lock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); } /** -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 134/206] ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
by Sasha Levin 18 Sep '20

18 Sep '20

From: Andreas Steinmetz <ast(a)domdv.de> [ Upstream commit 5c6cd7021a05a02fcf37f360592d7c18d4d807fb ] The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@do… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/midi.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 28a3ad8b1d74b..137e1e8718d6f 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1828,6 +1828,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1865,11 +1887,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 080/206] ALSA: usb-audio: Don't create a mixer element with bogus volume range
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e9a0ef0b5ddcbc0d56c65aefc0f18d16e6f71207 ] Some USB-audio descriptors provide a bogus volume range (e.g. volume min and max are identical), which confuses user-space. This patch makes the driver skipping such a control element. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206221 Link: https://lore.kernel.org/r/20200214144928.23628-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 45bd3d54be54b..451b8ea383c61 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1699,6 +1699,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, /* get min/max values */ get_min_max_with_quirks(cval, 0, kctl); + /* skip a bogus volume range */ + if (cval->max <= cval->min) { + usb_audio_dbg(mixer->chip, + "[%d] FU [%s] skipped due to invalid volume\n", + cval->head.id, kctl->id.name); + snd_ctl_free_one(kctl); + return; + } + + if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); if (cval->dBmin < cval->dBmax || !cval->initialized) { -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 070/206] ALSA: hda: Clear RIRB status before reading WP
by Sasha Levin 18 Sep '20

18 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 6d011d5057ff88ee556c000ac6fe0be23bdfcd72 ] RIRB interrupt status getting cleared after the write pointer is read causes a race condition, where last response(s) into RIRB may remain unserviced by IRQ, eventually causing azx_rirb_get_response to fall back to polling mode. Clearing the RIRB interrupt status ahead of write pointer access ensures that this condition is avoided. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Signed-off-by: Viswanath L <viswanathl(a)nvidia.com> Link: https://lore.kernel.org/r/1580983853-351-1-git-send-email-viswanathl@nvidia… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index fa261b27d8588..8198d2e53b7df 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1169,16 +1169,23 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update)) active = true; - /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures that no + * interrupt gets masked after the RIRB wp is read in + * snd_hdac_bus_update_rirb. This avoids a possible + * race condition where codec response in RIRB may + * remain unserviced by IRQ, eventually falling back + * to polling mode in azx_rirb_get_response. + */ + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); active = true; if (status & RIRB_INT_RESPONSE) { if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) udelay(80); snd_hdac_bus_update_rirb(bus); } - azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } } while (active && ++repeat < 10); -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 039/206] ASoC: max98090: remove msleep in PLL unlocked workaround
by Sasha Levin 18 Sep '20

18 Sep '20

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit acb874a7c049ec49d8fc66c893170fb42c01bdf7 ] It was observed Baytrail-based chromebooks could cause continuous PLL unlocked when using playback stream and capture stream simultaneously. Specifically, starting a capture stream after started a playback stream. As a result, the audio data could corrupt or turn completely silent. As the datasheet suggested, the maximum PLL lock time should be 7 msec. The workaround resets the codec softly by toggling SHDN off and on if PLL failed to lock for 10 msec. Notably, there is no suggested hold time for SHDN off. On Baytrail-based chromebooks, it would easily happen continuous PLL unlocked if there is a 10 msec delay between SHDN off and on. Removes the msleep(). Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20191122073114.219945-2-tzungbi@google.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 89b6e187ac235..a5b0c40ee545f 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2130,10 +2130,16 @@ static void max98090_pll_work(struct max98090_priv *max98090) dev_info_ratelimited(component->dev, "PLL unlocked\n"); + /* + * As the datasheet suggested, the maximum PLL lock time should be + * 7 msec. The workaround resets the codec softly by toggling SHDN + * off and on if PLL failed to lock for 10 msec. Notably, there is + * no suggested hold time for SHDN off. + */ + /* Toggle shutdown OFF then ON */ snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, 0); - msleep(10); snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, M98090_SHDNN_MASK); -- 2.25.1

1 0

[PATCH AUTOSEL 4.19 010/206] ASoC: kirkwood: fix IRQ error handling
by Sasha Levin 18 Sep '20

18 Sep '20

From: Russell King <rmk+kernel(a)armlinux.org.uk> [ Upstream commit 175fc928198236037174e5c5c066fe3c4691903e ] Propagate the error code from request_irq(), rather than returning -EBUSY. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> Link: https://lore.kernel.org/r/E1iNIqh-0000tW-EZ@rmk-PC.armlinux.org.uk Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index c6a58520d377a..255cc45905b81 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -136,7 +136,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 328/330] ALSA: hda: Workaround for spurious wakeups on some Intel platforms
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit a6630529aecb5a3e84370c376ed658e892e6261e ] We've received a regression report on Intel HD-audio controller that wakes up immediately after S3 suspend. The bisection leads to the commit c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed"). This commit replaces the system-suspend to use pm_runtime_force_suspend() instead of the direct call of __azx_runtime_suspend(). However, by some really mysterious reason, pm_runtime_force_suspend() causes a spurious wakeup (although it calls the same __azx_runtime_suspend() internally). As an ugly workaround for now, revert the behavior to call __azx_runtime_suspend() and __azx_runtime_resume() for those old Intel platforms that may exhibit such a problem, while keeping the new standard pm_runtime_force_suspend() and pm_runtime_force_resume() pair for the remaining chips. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208649 Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20200727164443.4233-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 17 ++++++++++++++--- 2 files changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 82e26442724ba..a356fb0e57738 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -/* 27 unused */ +#define AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP (1 << 27) /* Workaround for spurious wakeups after suspend */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 754e4d1a86b57..590ea262f2e20 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -295,7 +295,8 @@ enum { /* PCH for HSW/BDW; with runtime PM */ /* no i915 binding for this as HSW/BDW has another controller for HDMI */ #define AZX_DCAPS_INTEL_PCH \ - (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME) + (AZX_DCAPS_INTEL_PCH_BASE | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) /* HSW HDMI */ #define AZX_DCAPS_INTEL_HASWELL \ @@ -1026,7 +1027,14 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - pm_runtime_force_suspend(dev); + /* An ugly workaround: direct call of __azx_runtime_suspend() and + * __azx_runtime_resume() for old Intel platforms that suffer from + * spurious wakeups after S3 suspend + */ + if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + __azx_runtime_suspend(chip); + else + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1054,7 +1062,10 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - pm_runtime_force_resume(dev); + if (chip->driver_caps & AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP) + __azx_runtime_resume(chip, false); + else + pm_runtime_force_resume(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 327/330] ALSA: hda: Always use jackpoll helper for jack update after resume
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 8d6762af302d69f76fa788a277a56a9d9cd275d5 ] HD-audio codec driver applies a tricky procedure to forcibly perform the runtime resume by mimicking the usage count even if the device has been runtime-suspended beforehand. This was needed to assure to trigger the jack detection update after the system resume. And recently we also applied the similar logic to the HD-audio controller side. However this seems leading to some inconsistency, and eventually PCI controller gets screwed up. This patch is an attempt to fix and clean up those behavior: instead of the tricky runtime resume procedure, the existing jackpoll work is scheduled when such a forced codec resume is required. The jackpoll work will power up the codec, and this alone should suffice for the jack status update in usual cases. If the extra polling is requested (by checking codec->jackpoll_interval), the manual update is invoked after that, and the codec is powered down again. Also, we filter the spurious wake up of the codec from the controller runtime resume by checking codec->relaxed_resume flag. If this flag is set, basically we don't need to wake up explicitly, but it's supposed to be done via the audio component notifier. Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed") Link: https://lore.kernel.org/r/20200422203744.26299-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_codec.c | 28 +++++++++++++++++----------- sound/pci/hda/hda_intel.c | 17 ++--------------- 2 files changed, 19 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 12da263fb02ba..6da296def283e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -641,8 +641,18 @@ static void hda_jackpoll_work(struct work_struct *work) struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - snd_hda_jack_set_dirty_all(codec); - snd_hda_jack_poll_all(codec); + /* for non-polling trigger: we need nothing if already powered on */ + if (!codec->jackpoll_interval && snd_hdac_is_power_on(&codec->core)) + return; + + /* the power-up/down sequence triggers the runtime resume */ + snd_hda_power_up_pm(codec); + /* update jacks manually if polling is required, too */ + if (codec->jackpoll_interval) { + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_poll_all(codec); + } + snd_hda_power_down_pm(codec); if (!codec->jackpoll_interval) return; @@ -2958,18 +2968,14 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = hda_codec_need_resume(codec); int ret; - /* The get/put pair below enforces the runtime resume even if the - * device hasn't been used at suspend time. This trick is needed to - * update the jack state change during the sleep. - */ - if (forced_resume) - pm_runtime_get_noresume(dev); ret = pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); + /* schedule jackpoll work for jack detection update */ + if (codec->jackpoll_interval || + (pm_runtime_suspended(dev) && hda_codec_need_resume(codec))) + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); return ret; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a6e8aaa091c7d..754e4d1a86b57 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1002,7 +1002,8 @@ static void __azx_runtime_resume(struct azx *chip, bool from_rt) if (status && from_rt) { list_for_each_codec(codec, &chip->bus) - if (status & (1 << codec->addr)) + if (!codec->relaxed_resume && + (status & (1 << codec->addr))) schedule_delayed_work(&codec->jackpoll_work, codec->jackpoll_interval); } @@ -1041,9 +1042,7 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); - struct hda_codec *codec; struct azx *chip; - bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1055,19 +1054,7 @@ static int azx_resume(struct device *dev) if (azx_acquire_irq(chip, 1) < 0) return -EIO; - /* check for the forced resume */ - list_for_each_codec(codec, &chip->bus) { - if (hda_codec_need_resume(codec)) { - forced_resume = true; - break; - } - } - - if (forced_resume) - pm_runtime_get_noresume(dev); pm_runtime_force_resume(dev); - if (forced_resume) - pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 299/330] ASoC: img-i2s-out: Fix runtime PM imbalance on error
by Sasha Levin 18 Sep '20

18 Sep '20

From: Dinghao Liu <dinghao.liu(a)zju.edu.cn> [ Upstream commit 65bd91dd6957390c42a0491b9622cf31a2cdb140 ] pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Dinghao Liu <dinghao.liu(a)zju.edu.cn> Link: https://lore.kernel.org/r/20200529012230.5863-1-dinghao.liu@zju.edu.cn Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/img/img-i2s-out.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 4b18534096336..9c4212f2f7269 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -347,8 +347,10 @@ static int img_i2s_out_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) chan_control_mask = IMG_I2S_OUT_CHAN_CTL_CLKT_MASK; ret = pm_runtime_get_sync(i2s->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(i2s->dev); return ret; + } img_i2s_out_disable(i2s); @@ -488,8 +490,10 @@ static int img_i2s_out_probe(struct platform_device *pdev) goto err_pm_disable; } ret = pm_runtime_get_sync(&pdev->dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put_noidle(&pdev->dev); goto err_suspend; + } reg = IMG_I2S_OUT_CTL_FRM_SIZE_MASK; img_i2s_out_writel(i2s, reg, IMG_I2S_OUT_CTL); -- 2.25.1

1 0

[PATCH AUTOSEL 5.4 267/330] ALSA: hda: Fix potential race in unsol event handler
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c637fa151259c0f74665fde7cba5b7eac1417ae5 ] The unsol event handling code has a loop retrieving the read/write indices and the arrays without locking while the append to the array may happen concurrently. This may lead to some inconsistency. Although there hasn't been any proof of this bad results, it's still safer to protect the racy accesses. This patch adds the spinlock protection around the unsol handling loop for addressing it. Here we take bus->reg_lock as the writer side snd_hdac_bus_queue_event() is also protected by that lock. Link: https://lore.kernel.org/r/20200516062556.30951-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_bus.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 8f19876244ebe..53be2cac98e7c 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -158,6 +158,7 @@ static void snd_hdac_bus_process_unsol_events(struct work_struct *work) struct hdac_driver *drv; unsigned int rp, caddr, res; + spin_lock_irq(&bus->reg_lock); while (bus->unsol_rp != bus->unsol_wp) { rp = (bus->unsol_rp + 1) % HDA_UNSOL_QUEUE_SIZE; bus->unsol_rp = rp; @@ -169,10 +170,13 @@ static void snd_hdac_bus_process_unsol_events(struct work_struct *work) codec = bus->caddr_tbl[caddr & 0x0f]; if (!codec || !codec->dev.driver) continue; + spin_unlock_irq(&bus->reg_lock); drv = drv_to_hdac_driver(codec->dev.driver); if (drv->unsol_event) drv->unsol_event(codec, res); + spin_lock_irq(&bus->reg_lock); } + spin_unlock_irq(&bus->reg_lock); } /** -- 2.25.1

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[PATCH AUTOSEL 5.4 234/330] ALSA: hda: Skip controller resume if not needed
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c4c8dd6ef807663e42a5f04ea77cd62029eb99fa ] The HD-audio controller does system-suspend and resume operations by directly calling its helpers __azx_runtime_suspend() and __azx_runtime_resume(). However, in general, we don't have to resume always the device fully at the system resume; typically, if a device has been runtime-suspended, we can leave it to runtime resume. Usually for achieving this, the driver would call pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the system suspend and resume ops. Unfortunately, this doesn't work for the resume path in our case. For handling the jack detection at the system resume, a child codec device may need the (literally) forcibly resume even if it's been runtime-suspended, and for that, the controller device must be also resumed even if it's been suspended. This patch is an attempt to improve the situation. It replaces the direct __azx_runtime_suspend()/_resume() calls with with pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight trick as we've done for the codec side. More exactly: - azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and azx_runtime_resume(), so that it can be properly executed from the system-suspend/resume path - The WAKEEN handling depends on the card's power state now; it's set and cleared only for the runtime-suspend - azx_resume() checks whether any codec may need the forcible resume beforehand. If the forcible resume is required, it does temporary PM refcount up/down for actually triggering the runtime resume. - A new helper function, hda_codec_need_resume(), is introduced for checking whether the codec needs a forcible runtime-resume, and the existing code is rewritten with that. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043 Link: https://lore.kernel.org/r/20200413082034.25166-6-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/hda_codec.h | 5 +++++ sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/hda_intel.c | 38 +++++++++++++++++++++++++++----------- 3 files changed, 33 insertions(+), 12 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 9a0393cf024c2..65c056ce91128 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -494,6 +494,11 @@ void snd_hda_update_power_acct(struct hda_codec *codec); static inline void snd_hda_set_power_save(struct hda_bus *bus, int delay) {} #endif +static inline bool hda_codec_need_resume(struct hda_codec *codec) +{ + return !codec->relaxed_resume && codec->jacktbl.used; +} + #ifdef CONFIG_SND_HDA_PATCH_LOADER /* * patch firmware diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 103011e7285a3..12da263fb02ba 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2958,7 +2958,7 @@ static int hda_codec_runtime_resume(struct device *dev) static int hda_codec_force_resume(struct device *dev) { struct hda_codec *codec = dev_to_hda_codec(dev); - bool forced_resume = !codec->relaxed_resume && codec->jacktbl.used; + bool forced_resume = hda_codec_need_resume(codec); int ret; /* The get/put pair below enforces the runtime resume even if the diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7353d2ec359ae..a6e8aaa091c7d 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1025,7 +1025,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - __azx_runtime_suspend(chip); + pm_runtime_force_suspend(dev); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1041,7 +1041,9 @@ static int azx_suspend(struct device *dev) static int azx_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); + struct hda_codec *codec; struct azx *chip; + bool forced_resume = false; if (!azx_is_pm_ready(card)) return 0; @@ -1052,7 +1054,20 @@ static int azx_resume(struct device *dev) chip->msi = 0; if (azx_acquire_irq(chip, 1) < 0) return -EIO; - __azx_runtime_resume(chip, false); + + /* check for the forced resume */ + list_for_each_codec(codec, &chip->bus) { + if (hda_codec_need_resume(codec)) { + forced_resume = true; + break; + } + } + + if (forced_resume) + pm_runtime_get_noresume(dev); + pm_runtime_force_resume(dev); + if (forced_resume) + pm_runtime_put(dev); snd_power_change_state(card, SNDRV_CTL_POWER_D0); trace_azx_resume(chip); @@ -1099,12 +1114,12 @@ static int azx_runtime_suspend(struct device *dev) if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; /* enable controller wake up event */ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | - STATESTS_INT_MASK); + if (snd_power_get_state(card) == SNDRV_CTL_POWER_D0) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + } __azx_runtime_suspend(chip); trace_azx_runtime_suspend(chip); @@ -1115,17 +1130,18 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip; + bool from_rt = snd_power_get_state(card) == SNDRV_CTL_POWER_D0; if (!azx_is_pm_ready(card)) return 0; chip = card->private_data; - if (!azx_has_pm_runtime(chip)) - return 0; - __azx_runtime_resume(chip, true); + __azx_runtime_resume(chip, from_rt); /* disable controller Wake Up event*/ - azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & - ~STATESTS_INT_MASK); + if (from_rt) { + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); + } trace_azx_runtime_resume(chip); return 0; -- 2.25.1

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[PATCH AUTOSEL 5.4 220/330] ALSA: usb-audio: Fix case when USB MIDI interface has more than one extra endpoint descriptor
by Sasha Levin 18 Sep '20

18 Sep '20

From: Andreas Steinmetz <ast(a)domdv.de> [ Upstream commit 5c6cd7021a05a02fcf37f360592d7c18d4d807fb ] The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra endpoint descriptor. The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00 As the code in snd_usbmidi_get_ms_info() looks only at the first extra descriptor to find USB_DT_CS_ENDPOINT the device as such is recognized but there is neither input nor output configured. The patch iterates through the extra descriptors to find the proper one. With this patch the device is correctly configured. Signed-off-by: Andreas Steinmetz <ast(a)domdv.de> Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@do… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/midi.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 0cb4142b05f64..bc9068b616bb9 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1827,6 +1827,28 @@ static int snd_usbmidi_create_endpoints(struct snd_usb_midi *umidi, return 0; } +static struct usb_ms_endpoint_descriptor *find_usb_ms_endpoint_descriptor( + struct usb_host_endpoint *hostep) +{ + unsigned char *extra = hostep->extra; + int extralen = hostep->extralen; + + while (extralen > 3) { + struct usb_ms_endpoint_descriptor *ms_ep = + (struct usb_ms_endpoint_descriptor *)extra; + + if (ms_ep->bLength > 3 && + ms_ep->bDescriptorType == USB_DT_CS_ENDPOINT && + ms_ep->bDescriptorSubtype == UAC_MS_GENERAL) + return ms_ep; + if (!extra[0]) + break; + extralen -= extra[0]; + extra += extra[0]; + } + return NULL; +} + /* * Returns MIDIStreaming device capabilities. */ @@ -1864,11 +1886,8 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi *umidi, ep = get_ep_desc(hostep); if (!usb_endpoint_xfer_bulk(ep) && !usb_endpoint_xfer_int(ep)) continue; - ms_ep = (struct usb_ms_endpoint_descriptor *)hostep->extra; - if (hostep->extralen < 4 || - ms_ep->bLength < 4 || - ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) + ms_ep = find_usb_ms_endpoint_descriptor(hostep); + if (!ms_ep) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { -- 2.25.1

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[PATCH AUTOSEL 5.4 149/330] ASoC: SOF: ipc: check ipc return value before data copy
by Sasha Levin 18 Sep '20

18 Sep '20

From: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> [ Upstream commit 1919b42ca4ad75a2397081164661af3ce5a7b8f4 ] In tx_wait_done the ipc payload is copied before the DSP transaction error code is checked. This might lead to corrupted data in kernel side even though the error would be handled later. It is also pointless to copy the data in case of error. So change the order of error check and copy. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200228231850.9226-3-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/ipc.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index e7b1a80e2a14c..f38f651da2246 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -215,15 +215,17 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg, snd_sof_trace_notify_for_error(ipc->sdev); ret = -ETIMEDOUT; } else { - /* copy the data returned from DSP */ ret = msg->reply_error; - if (msg->reply_size) - memcpy(reply_data, msg->reply_data, msg->reply_size); - if (ret < 0) + if (ret < 0) { dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n", hdr->cmd, msg->reply_size); - else + } else { ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); + if (msg->reply_size) + /* copy the data returned from DSP */ + memcpy(reply_data, msg->reply_data, + msg->reply_size); + } } return ret; -- 2.25.1

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[PATCH AUTOSEL 5.4 131/330] ALSA: usb-audio: Don't create a mixer element with bogus volume range
by Sasha Levin 18 Sep '20

18 Sep '20

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit e9a0ef0b5ddcbc0d56c65aefc0f18d16e6f71207 ] Some USB-audio descriptors provide a bogus volume range (e.g. volume min and max are identical), which confuses user-space. This patch makes the driver skipping such a control element. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206221 Link: https://lore.kernel.org/r/20200214144928.23628-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/mixer.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9079c380228fc..8aa96ed0b1b56 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1684,6 +1684,16 @@ static void __build_feature_ctl(struct usb_mixer_interface *mixer, /* get min/max values */ get_min_max_with_quirks(cval, 0, kctl); + /* skip a bogus volume range */ + if (cval->max <= cval->min) { + usb_audio_dbg(mixer->chip, + "[%d] FU [%s] skipped due to invalid volume\n", + cval->head.id, kctl->id.name); + snd_ctl_free_one(kctl); + return; + } + + if (control == UAC_FU_VOLUME) { check_mapped_dB(map, cval); if (cval->dBmin < cval->dBmax || !cval->initialized) { -- 2.25.1

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[PATCH AUTOSEL 5.4 116/330] ALSA: hda: Clear RIRB status before reading WP
by Sasha Levin 18 Sep '20

18 Sep '20

From: Mohan Kumar <mkumard(a)nvidia.com> [ Upstream commit 6d011d5057ff88ee556c000ac6fe0be23bdfcd72 ] RIRB interrupt status getting cleared after the write pointer is read causes a race condition, where last response(s) into RIRB may remain unserviced by IRQ, eventually causing azx_rirb_get_response to fall back to polling mode. Clearing the RIRB interrupt status ahead of write pointer access ensures that this condition is avoided. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> Signed-off-by: Viswanath L <viswanathl(a)nvidia.com> Link: https://lore.kernel.org/r/1580983853-351-1-git-send-email-viswanathl@nvidia… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_controller.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 76b507058cb4d..5e6081750bd9b 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1159,16 +1159,23 @@ irqreturn_t azx_interrupt(int irq, void *dev_id) if (snd_hdac_bus_handle_stream_irq(bus, status, stream_update)) active = true; - /* clear rirb int */ status = azx_readb(chip, RIRBSTS); if (status & RIRB_INT_MASK) { + /* + * Clearing the interrupt status here ensures that no + * interrupt gets masked after the RIRB wp is read in + * snd_hdac_bus_update_rirb. This avoids a possible + * race condition where codec response in RIRB may + * remain unserviced by IRQ, eventually falling back + * to polling mode in azx_rirb_get_response. + */ + azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); active = true; if (status & RIRB_INT_RESPONSE) { if (chip->driver_caps & AZX_DCAPS_CTX_WORKAROUND) udelay(80); snd_hdac_bus_update_rirb(bus); } - azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); } } while (active && ++repeat < 10); -- 2.25.1

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[PATCH AUTOSEL 5.4 077/330] ALSA: hda: enable regmap internal locking
by Sasha Levin 18 Sep '20

18 Sep '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 8e85def5723eccea30ebf22645673692ab8cb3e2 ] This reverts commit 42ec336f1f9d ("ALSA: hda: Disable regmap internal locking"). Without regmap locking, there is a race between snd_hda_codec_amp_init() and PM callbacks issuing regcache_sync(). This was caught by following kernel warning trace: <4> [358.080081] WARNING: CPU: 2 PID: 4157 at drivers/base/regmap/regcache.c:498 regcache_cache_only+0xf5/0x130 [...] <4> [358.080148] Call Trace: <4> [358.080158] snd_hda_codec_amp_init+0x4e/0x100 [snd_hda_codec] <4> [358.080169] snd_hda_codec_amp_init_stereo+0x40/0x80 [snd_hda_codec] Suggested-by: Takashi Iwai <tiwai(a)suse.de> BugLink: https://gitlab.freedesktop.org/drm/intel/issues/592 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Link: https://lore.kernel.org/r/20200108180856.5194-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_regmap.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 2596a881186fa..49780399c2849 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -363,7 +363,6 @@ static const struct regmap_config hda_regmap_cfg = { .reg_write = hda_reg_write, .use_single_read = true, .use_single_write = true, - .disable_locking = true, }; /** -- 2.25.1

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[PATCH AUTOSEL 5.4 065/330] ASoC: max98090: remove msleep in PLL unlocked workaround
by Sasha Levin 18 Sep '20

18 Sep '20

From: Tzung-Bi Shih <tzungbi(a)google.com> [ Upstream commit acb874a7c049ec49d8fc66c893170fb42c01bdf7 ] It was observed Baytrail-based chromebooks could cause continuous PLL unlocked when using playback stream and capture stream simultaneously. Specifically, starting a capture stream after started a playback stream. As a result, the audio data could corrupt or turn completely silent. As the datasheet suggested, the maximum PLL lock time should be 7 msec. The workaround resets the codec softly by toggling SHDN off and on if PLL failed to lock for 10 msec. Notably, there is no suggested hold time for SHDN off. On Baytrail-based chromebooks, it would easily happen continuous PLL unlocked if there is a 10 msec delay between SHDN off and on. Removes the msleep(). Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> Link: https://lore.kernel.org/r/20191122073114.219945-2-tzungbi@google.com Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98090.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 45da2b51543e7..6b9d326e11b07 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2112,10 +2112,16 @@ static void max98090_pll_work(struct max98090_priv *max98090) dev_info_ratelimited(component->dev, "PLL unlocked\n"); + /* + * As the datasheet suggested, the maximum PLL lock time should be + * 7 msec. The workaround resets the codec softly by toggling SHDN + * off and on if PLL failed to lock for 10 msec. Notably, there is + * no suggested hold time for SHDN off. + */ + /* Toggle shutdown OFF then ON */ snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, 0); - msleep(10); snd_soc_component_update_bits(component, M98090_REG_DEVICE_SHUTDOWN, M98090_SHDNN_MASK, M98090_SHDNN_MASK); -- 2.25.1

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[PATCH AUTOSEL 5.4 019/330] ASoC: kirkwood: fix IRQ error handling
by Sasha Levin 18 Sep '20

18 Sep '20

From: Russell King <rmk+kernel(a)armlinux.org.uk> [ Upstream commit 175fc928198236037174e5c5c066fe3c4691903e ] Propagate the error code from request_irq(), rather than returning -EBUSY. Signed-off-by: Russell King <rmk+kernel(a)armlinux.org.uk> Link: https://lore.kernel.org/r/E1iNIqh-0000tW-EZ@rmk-PC.armlinux.org.uk Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/kirkwood/kirkwood-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index 6f69f314f2c2a..d2d5c25bf5502 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -132,7 +132,7 @@ static int kirkwood_dma_open(struct snd_pcm_substream *substream) err = request_irq(priv->irq, kirkwood_dma_irq, IRQF_SHARED, "kirkwood-i2s", priv); if (err) - return -EBUSY; + return err; /* * Enable Error interrupts. We're only ack'ing them but -- 2.25.1

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[PATCH v2] ASoC: dt-bindings: correct interrupt flags in examples
by Krzysztof Kozlowski 17 Sep '20

17 Sep '20

GPIO_ACTIVE_x flags are not correct in the context of interrupt flags. These are simple defines so they could be used in DTS but they will not have the same meaning: 1. GPIO_ACTIVE_HIGH = 0 = IRQ_TYPE_NONE 2. GPIO_ACTIVE_LOW = 1 = IRQ_TYPE_EDGE_RISING Correct the interrupt flags, assuming the author of the code wanted same logical behavior behind the name "ACTIVE_xxx", this is: ACTIVE_HIGH => IRQ_TYPE_LEVEL_HIGH Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Rob Herring <robh(a)kernel.org> --- Changes since v1: 1. Add acks --- Documentation/devicetree/bindings/sound/max98090.txt | 2 +- Documentation/devicetree/bindings/sound/rt5640.txt | 2 +- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- Documentation/devicetree/bindings/sound/rt5665.txt | 2 +- Documentation/devicetree/bindings/sound/rt5668.txt | 2 +- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index 7e1bbd5c27fd..39d640294c62 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -55,5 +55,5 @@ audio-codec@10 { compatible = "maxim,max98090"; reg = <0x10>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(H, 4) IRQ_TYPE_LEVEL_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index e40e4893eed8..ff1228713f7e 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -88,7 +88,7 @@ rt5640 { compatible = "realtek,rt5640"; reg = <0x1c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 1766e0543fc5..56788f50b6cf 100644 --- a/Documentation/devicetree/bindings/sound/rt5659.txt +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -72,7 +72,7 @@ rt5659 { compatible = "realtek,rt5659"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt index 8df170506986..f6ca96b4ce98 100644 --- a/Documentation/devicetree/bindings/sound/rt5665.txt +++ b/Documentation/devicetree/bindings/sound/rt5665.txt @@ -62,7 +62,7 @@ rt5659 { compatible = "realtek,rt5665"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt index c88b96e7764b..a2b7e9a2f2f3 100644 --- a/Documentation/devicetree/bindings/sound/rt5668.txt +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -41,7 +41,7 @@ rt5668 { compatible = "realtek,rt5668b"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 1b3c13d206ff..da2430099181 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -64,7 +64,7 @@ rt5677 { compatible = "realtek,rt5677"; reg = <0x2c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; gpio-controller; #gpio-cells = <2>; diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index ade1ece8b45f..707fa98d1310 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -58,7 +58,7 @@ rt5682 { compatible = "realtek,rt5682i"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; -- 2.17.1

2 1

Re: [trivial PATCH] treewide: Convert switch/case fallthrough; to break;
by Jacob Keller 17 Sep '20

17 Sep '20

On 9/9/2020 1:55 PM, Keith Busch wrote: > On Wed, Sep 09, 2020 at 01:06:39PM -0700, Joe Perches wrote: >> diff --git a/crypto/tcrypt.c b/crypto/tcrypt.c >> index eea0f453cfb6..8aac5bc60f4c 100644 >> --- a/crypto/tcrypt.c >> +++ b/crypto/tcrypt.c >> @@ -2464,7 +2464,7 @@ static int do_test(const char *alg, u32 type, u32 mask, int m, u32 num_mb) >> test_hash_speed("streebog512", sec, >> generic_hash_speed_template); >> if (mode > 300 && mode < 400) break; >> - fallthrough; >> + break; >> case 399: >> break; > > Just imho, this change makes the preceding 'if' look even more > pointless. Maybe the fallthrough was a deliberate choice? Not that my > opinion matters here as I don't know this module, but it looked a bit > odd to me. > Yea this does look very odd..

1 0

[PATCH v2 0/3] ASoC: fsl_sai: update the register list
by Shengjiu Wang 17 Sep '20

17 Sep '20

As sai ip is upgraded, so update sai register list. Shengjiu Wang (3): ASoC: fsl_sai: Add new added registers and new bit definition ASoC: fsl_sai: Add fsl_sai_check_version function ASoC: fsl_sai: Set MCLK input or output direction changes in v2: - update commit message for first commit - Add acked-by Nicolin sound/soc/fsl/fsl_sai.c | 77 ++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 87 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 164 insertions(+) -- 2.27.0

3 5

Re: [PATCH] ASoC: fsl_audmix: make clock and output src write only
by Mark Brown 17 Sep '20

17 Sep '20

On Mon, 14 Sep 2020 20:24:34 +0300, Viorel Suman (OSS) wrote: > "alsactl -f state.conf store/restore" sequence fails because setting > "mixing clock source" and "output source" requires active TDM clock > being started for configuration propagation. Make these two controls > write only so that their values are not stored at "alsactl store". Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: fsl_audmix: make clock and output src write only commit: 944c517b8c838832a166f1c89afbf8724f4a6b49 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[PATCH] ASoC: dt-bindings: aries-wm8994: Match compatibles with enum
by Krzysztof Kozlowski 17 Sep '20

17 Sep '20

The common pattern for enumerating compatibles is enum, not oneOf. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- .../bindings/sound/samsung,aries-wm8994.yaml | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml index 902a0b66628e..1c6947294825 100644 --- a/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,aries-wm8994.yaml @@ -11,12 +11,11 @@ maintainers: properties: compatible: - oneOf: - - const: samsung,aries-wm8994 - description: With FM radio and modem master - - - const: samsung,fascinate4g-wm8994 - description: Without FM radio and modem slave + enum: + # With FM radio and modem master + - samsung,aries-wm8994 + # Without FM radio and modem slave + - samsung,fascinate4g-wm8994 model: $ref: /schemas/types.yaml#/definitions/string -- 2.17.1

3 2

[PATCH 1/6] ASoC: tlv320adcx140: Idle the device while writing registers
by Dan Murphy 17 Sep '20

17 Sep '20

It was observed that if the device was active and register writes were performed there were some unwanted behaviors particularly when writing the word length and some filter options. So when writing to the device the device should be placed in sleep mode and then exit sleep mode once the register update is complete. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 43 ++++++++++++++++++++++++-------- sound/soc/codecs/tlv320adcx140.h | 2 ++ 2 files changed, 34 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 8efe20605f9b..3909c1cf52be 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -30,7 +30,7 @@ struct adcx140_priv { struct regmap *regmap; struct device *dev; - int micbias_vg; + bool micbias_vg; unsigned int dai_fmt; unsigned int tdm_delay; @@ -614,11 +614,26 @@ static int adcx140_reset(struct adcx140_priv *adcx140) return ret; } +static void adcx140_pwr_ctrl(struct adcx140_priv *adcx140, bool power_state) +{ + int pwr_ctrl = 0; + + if (power_state) + pwr_ctrl = ADCX140_PWR_CFG_ADC_PDZ | ADCX140_PWR_CFG_PLL_PDZ; + + if (adcx140->micbias_vg && power_state) + pwr_ctrl |= ADCX140_PWR_CFG_BIAS_PDZ; + + regmap_update_bits(adcx140->regmap, ADCX140_PWR_CFG, + ADCX140_PWR_CTRL_MSK, pwr_ctrl); +} + static int adcx140_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); u8 data = 0; switch (params_width(params)) { @@ -640,9 +655,13 @@ static int adcx140_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + adcx140_pwr_ctrl(adcx140, false); + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, ADCX140_WORD_LEN_MSK, data); + adcx140_pwr_ctrl(adcx140, true); + return 0; } @@ -709,6 +728,8 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, adcx140->dai_fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + adcx140_pwr_ctrl(adcx140, false); + snd_soc_component_update_bits(component, ADCX140_ASI_CFG0, ADCX140_FSYNCINV_BIT | ADCX140_BCLKINV_BIT | @@ -721,6 +742,7 @@ static int adcx140_set_dai_fmt(struct snd_soc_dai *codec_dai, snd_soc_component_update_bits(component, ADCX140_ASI_CFG1, ADCX140_TX_OFFSET_MASK, offset); + adcx140_pwr_ctrl(adcx140, true); return 0; } @@ -818,12 +840,11 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ret = device_property_read_u32(adcx140->dev, "ti,mic-bias-source", &bias_source); - if (ret) + if (ret || bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { bias_source = ADCX140_MIC_BIAS_VAL_VREF; - - if (bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { - dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); - return -EINVAL; + adcx140->micbias_vg = false; + } else { + adcx140->micbias_vg = true; } ret = device_property_read_u32(adcx140->dev, "ti,vref-source", @@ -906,6 +927,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ADCX140_MIC_BIAS_VREF_MSK, bias_cfg); if (ret) dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret); + + adcx140_pwr_ctrl(adcx140, true); out: return ret; } @@ -914,21 +937,19 @@ static int adcx140_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { struct adcx140_priv *adcx140 = snd_soc_component_get_drvdata(component); - int pwr_cfg = 0; switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: case SND_SOC_BIAS_STANDBY: - pwr_cfg = ADCX140_PWR_CFG_BIAS_PDZ | ADCX140_PWR_CFG_PLL_PDZ | - ADCX140_PWR_CFG_ADC_PDZ; + adcx140_pwr_ctrl(adcx140, true); break; case SND_SOC_BIAS_OFF: - pwr_cfg = 0x0; + adcx140_pwr_ctrl(adcx140, false); break; } - return regmap_write(adcx140->regmap, ADCX140_PWR_CFG, pwr_cfg); + return 0; } static const struct snd_soc_component_driver soc_codec_driver_adcx140 = { diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index eedbc1d7221f..94c6d1fd2977 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -123,6 +123,7 @@ #define ADCX140_MIC_BIAS_VREF_1375V 2 #define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0) +#define ADCX140_PWR_CTRL_MSK GENMASK(7, 5) #define ADCX140_PWR_CFG_BIAS_PDZ BIT(7) #define ADCX140_PWR_CFG_ADC_PDZ BIT(6) #define ADCX140_PWR_CFG_PLL_PDZ BIT(5) @@ -145,4 +146,5 @@ #define ADCX140_GPO_CFG_MAX 4 #define ADCX140_GPO_DRV_MAX 5 + #endif /* _TLV320ADCX140_ */ -- 2.28.0

2 9

[PATCH] ASoC: qcom: lpass-sc7180: Add MODULE_DEVICE_TABLE
by Douglas Anderson 17 Sep '20

17 Sep '20

The lpass-sc7180 driver can be built as a module but is lacking a MODULE_DEVICE_TABLE. This means it won't auto-load. Fix this oversight. Fixes: 24caf8d9eb10 ("ASoC: qcom: lpass-sc7180: Add platform driver for lpass audio") Signed-off-by: Douglas Anderson <dianders(a)chromium.org> --- sound/soc/qcom/lpass-sc7180.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index 167bf2cbc556..a8a3d8f8f567 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -200,6 +200,7 @@ static const struct of_device_id sc7180_lpass_cpu_device_id[] = { {.compatible = "qcom,sc7180-lpass-cpu", .data = &sc7180_data}, {} }; +MODULE_DEVICE_TABLE(of, sc7180_lpass_cpu_device_id); static struct platform_driver sc7180_lpass_cpu_platform_driver = { .driver = { -- 2.28.0.618.gf4bc123cb7-goog

2 1

[PATCH] ASoC: dt-bindings: Correct interrupt flags in examples
by Krzysztof Kozlowski 17 Sep '20

17 Sep '20

GPIO_ACTIVE_x flags are not correct in the context of interrupt flags. These are simple defines so they could be used in DTS but they will not have the same meaning: 1. GPIO_ACTIVE_HIGH = 0 = IRQ_TYPE_NONE 2. GPIO_ACTIVE_LOW = 1 = IRQ_TYPE_EDGE_RISING Correct the interrupt flags, assuming the author of the code wanted some logical behavior behind the name "ACTIVE_xxx", this is: ACTIVE_HIGH => IRQ_TYPE_LEVEL_HIGH Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Documentation/devicetree/bindings/sound/max98090.txt | 2 +- Documentation/devicetree/bindings/sound/rt5640.txt | 2 +- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- Documentation/devicetree/bindings/sound/rt5665.txt | 2 +- Documentation/devicetree/bindings/sound/rt5668.txt | 2 +- Documentation/devicetree/bindings/sound/rt5677.txt | 2 +- Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- 7 files changed, 7 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index 7e1bbd5c27fd..39d640294c62 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -55,5 +55,5 @@ audio-codec@10 { compatible = "maxim,max98090"; reg = <0x10>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(H, 4) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(H, 4) IRQ_TYPE_LEVEL_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index e40e4893eed8..ff1228713f7e 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -88,7 +88,7 @@ rt5640 { compatible = "realtek,rt5640"; reg = <0x1c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 1766e0543fc5..56788f50b6cf 100644 --- a/Documentation/devicetree/bindings/sound/rt5659.txt +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -72,7 +72,7 @@ rt5659 { compatible = "realtek,rt5659"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5665.txt b/Documentation/devicetree/bindings/sound/rt5665.txt index 8df170506986..f6ca96b4ce98 100644 --- a/Documentation/devicetree/bindings/sound/rt5665.txt +++ b/Documentation/devicetree/bindings/sound/rt5665.txt @@ -62,7 +62,7 @@ rt5659 { compatible = "realtek,rt5665"; reg = <0x1b>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; }; diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt index c88b96e7764b..a2b7e9a2f2f3 100644 --- a/Documentation/devicetree/bindings/sound/rt5668.txt +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -41,7 +41,7 @@ rt5668 { compatible = "realtek,rt5668b"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; diff --git a/Documentation/devicetree/bindings/sound/rt5677.txt b/Documentation/devicetree/bindings/sound/rt5677.txt index 1b3c13d206ff..da2430099181 100644 --- a/Documentation/devicetree/bindings/sound/rt5677.txt +++ b/Documentation/devicetree/bindings/sound/rt5677.txt @@ -64,7 +64,7 @@ rt5677 { compatible = "realtek,rt5677"; reg = <0x2c>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(W, 3) IRQ_TYPE_LEVEL_HIGH>; gpio-controller; #gpio-cells = <2>; diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index ade1ece8b45f..707fa98d1310 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -58,7 +58,7 @@ rt5682 { compatible = "realtek,rt5682i"; reg = <0x1a>; interrupt-parent = <&gpio>; - interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + interrupts = <TEGRA_GPIO(U, 6) IRQ_TYPE_LEVEL_HIGH>; realtek,ldo1-en-gpios = <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; realtek,dmic1-data-pin = <1>; -- 2.17.1

3 2

[PATCH] ASoC: tlv320adcx140: Fix digital gain range
by Camel Guo 17 Sep '20

17 Sep '20

From: Camel Guo <camelg(a)axis.com> According to its datasheet, the digital gain should be -100 dB when CHx_DVOL is 1 and 27 dB when CHx_DVOL is 255. But with the current dig_vol_tlv, "Digital CHx Out Volume" shows 27.5 dB if CHx_DVOL is 255 and -95.5 dB if CHx_DVOL is 1. This commit fixes this bug. Fixes: 689c7655b50c ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Camel Guo <camelg(a)axis.com> --- sound/soc/codecs/tlv320adcx140.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 6d456aa269ad..13ece039e9fe 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -161,7 +161,7 @@ static const struct regmap_config adcx140_i2c_regmap = { }; /* Digital Volume control. From -100 to 27 dB in 0.5 dB steps */ -static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); +static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10050, 50, 0); /* ADC gain. From 0 to 42 dB in 1 dB steps */ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); -- 2.20.1

2 1

[PATCH v3 1/2] ASoC: dt-bindings: ak4458: Add dsd-path property
by Shengjiu Wang 17 Sep '20

17 Sep '20

Add "dsd-path" property, which is used for ak4497 codec to select the DSD input pin. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Documentation/devicetree/bindings/sound/ak4458.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/ak4458.txt b/Documentation/devicetree/bindings/sound/ak4458.txt index 8f6c84f21468..0416c14895d6 100644 --- a/Documentation/devicetree/bindings/sound/ak4458.txt +++ b/Documentation/devicetree/bindings/sound/ak4458.txt @@ -12,6 +12,9 @@ Optional properties: - mute-gpios: A GPIO specifier for the soft mute pin - AVDD-supply: Analog power supply - DVDD-supply: Digital power supply +- dsd-path: Select DSD input pins for ak4497 + 0: select #16, #17, #19 pins + 1: select #3, #4, #5 pins Example: -- 2.27.0

2 2

[PATCH 0/2] ASoC: q6afe: add clocks support
by Srinivas Kandagatla 17 Sep '20

17 Sep '20

q6afe already exposes clocks using apis, but not as proper clock controller driver. This patch puts those clocks in to a proper clock controller so that other drivers that depend on those clocks can be properly expressed. Srinivas Kandagatla (2): ASoC: q6afe: dt-bindings: add q6afe clock bindings ASoC: q6afe-clocks: add q6afe clock controller .../devicetree/bindings/sound/qcom,q6afe.txt | 23 ++ include/dt-bindings/sound/qcom,q6afe.h | 74 ++++- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-clocks.c | 270 ++++++++++++++++++ 5 files changed, 371 insertions(+), 1 deletion(-) create mode 100644 sound/soc/qcom/qdsp6/q6afe-clocks.c -- 2.21.0

2 3

[PATCH 0/8] ASoC: SOF: small fixes for 5.10
by Kai Vehmanen 17 Sep '20

17 Sep '20

Series that adds debug support for IMX platforms, more details to FW version information, adds missing -EACCESS handling to pm_runtime_get_sync() calls and a set of minor cosmetic, trace verbosity and coding style issues. Guennadi Liakhovetski (3): ASoC: SOF: (cosmetic) remove redundant "ret" variable uses ASoC: SOF: remove several superfluous type-casts ASoC: SOF: fix range checks Iulian Olaru (1): ASoC: SOF: imx: Add debug support for imx platforms Karol Trzcinski (1): ASoC: SOF: Add `src_hash` to `sof_ipc_fw_version` structure Pierre-Louis Bossart (3): ASoC: SOF: debug: update test for pm_runtime_get_sync() ASoC: SOF: control: update test for pm_runtime_get_sync() ASoC: SOF: Intel: hda: reduce verbosity of boot error logs include/sound/sof/info.h | 4 +- sound/soc/sof/control.c | 62 +++++++++++++-------------- sound/soc/sof/debug.c | 2 +- sound/soc/sof/imx/Kconfig | 8 ++++ sound/soc/sof/imx/Makefile | 3 ++ sound/soc/sof/imx/imx-common.c | 72 ++++++++++++++++++++++++++++++++ sound/soc/sof/imx/imx-common.h | 16 +++++++ sound/soc/sof/imx/imx8.c | 23 +++++++++- sound/soc/sof/imx/imx8m.c | 17 +++++++- sound/soc/sof/intel/hda-loader.c | 16 +++---- sound/soc/sof/intel/hda.c | 12 ++++-- sound/soc/sof/intel/hda.h | 2 + sound/soc/sof/sof-audio.c | 6 +-- sound/soc/sof/sof-priv.h | 8 ++++ sound/soc/sof/topology.c | 44 ++++++++++--------- 15 files changed, 226 insertions(+), 69 deletions(-) create mode 100644 sound/soc/sof/imx/imx-common.c create mode 100644 sound/soc/sof/imx/imx-common.h -- 2.27.0

2 9

[PATCH] ASoC: SOF: Intel: Add support for tgl-h
by Kai Vehmanen 17 Sep '20

17 Sep '20

From: Rander Wang <rander.wang(a)intel.com> SOF will support tgl-h and tgl-lp in different FW binaries due to hardware difference, so create another dev_desc entry with FW name of sof-tgl-h.ri and dsp_desc named tglh_chip_info for tgl-h. Fixes: c8d2e2bfaeffa ("ASoC: SOF: Intel: add PCI IDs for ICL-H and TGL-H") Signed-off-by: Rander Wang <rander.wang(a)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/intel/hda.h | 1 + sound/soc/sof/intel/tgl.c | 16 ++++++++++++++++ sound/soc/sof/sof-pci-dev.c | 18 +++++++++++++++++- 3 files changed, 34 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index f0f8f95c082b..659868da4068 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -737,6 +737,7 @@ extern const struct sof_intel_dsp_desc cnl_chip_info; extern const struct sof_intel_dsp_desc skl_chip_info; extern const struct sof_intel_dsp_desc icl_chip_info; extern const struct sof_intel_dsp_desc tgl_chip_info; +extern const struct sof_intel_dsp_desc tglh_chip_info; extern const struct sof_intel_dsp_desc ehl_chip_info; extern const struct sof_intel_dsp_desc jsl_chip_info; diff --git a/sound/soc/sof/intel/tgl.c b/sound/soc/sof/intel/tgl.c index f8d04fd66ceb..0278b67de1ec 100644 --- a/sound/soc/sof/intel/tgl.c +++ b/sound/soc/sof/intel/tgl.c @@ -135,3 +135,19 @@ const struct sof_intel_dsp_desc tgl_chip_info = { .ssp_base_offset = CNL_SSP_BASE_OFFSET, }; EXPORT_SYMBOL_NS(tgl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); + +const struct sof_intel_dsp_desc tglh_chip_info = { + /* Tigerlake-H */ + .cores_num = 2, + .init_core_mask = 1, + .host_managed_cores_mask = BIT(0), + .ipc_req = CNL_DSP_REG_HIPCIDR, + .ipc_req_mask = CNL_DSP_REG_HIPCIDR_BUSY, + .ipc_ack = CNL_DSP_REG_HIPCIDA, + .ipc_ack_mask = CNL_DSP_REG_HIPCIDA_DONE, + .ipc_ctl = CNL_DSP_REG_HIPCCTL, + .rom_init_timeout = 300, + .ssp_count = ICL_SSP_COUNT, + .ssp_base_offset = CNL_SSP_BASE_OFFSET, +}; +EXPORT_SYMBOL_NS(tglh_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 3ec380945466..8c53f6935417 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -223,6 +223,22 @@ static const struct sof_dev_desc tgl_desc = { .nocodec_tplg_filename = "sof-tgl-nocodec.tplg", .ops = &sof_tgl_ops, }; + +static const struct sof_dev_desc tglh_desc = { + .machines = snd_soc_acpi_intel_tgl_machines, + .alt_machines = snd_soc_acpi_intel_tgl_sdw_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = -1, + .resindex_imr_base = -1, + .irqindex_host_ipc = -1, + .resindex_dma_base = -1, + .chip_info = &tglh_chip_info, + .default_fw_path = "intel/sof", + .default_tplg_path = "intel/sof-tplg", + .default_fw_filename = "sof-tgl-h.ri", + .nocodec_tplg_filename = "sof-tgl-nocodec.tplg", + .ops = &sof_tgl_ops, +}; #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) @@ -457,7 +473,7 @@ static const struct pci_device_id sof_pci_ids[] = { { PCI_DEVICE(0x8086, 0xa0c8), /* TGL-LP */ .driver_data = (unsigned long)&tgl_desc}, { PCI_DEVICE(0x8086, 0x43c8), /* TGL-H */ - .driver_data = (unsigned long)&tgl_desc}, + .driver_data = (unsigned long)&tglh_desc}, #endif #if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) -- 2.27.0

2 1

[PATCH 1/2] ASoC: codec: tlv320aic32x4: fix missing aic32x4_disable_regulators() in error path
by Matthias Schiffer 17 Sep '20

17 Sep '20

The regulators need to be disabled in the aic32x4_register_clocks() failure case as well. Fixes: 9d4befff5a95 ("ASoC: codec: tlv3204: Moving GPIO reset and add ADC reset") Signed-off-by: Matthias Schiffer <matthias.schiffer(a)ew.tq-group.com> --- sound/soc/codecs/tlv320aic32x4.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 8dcea566b375..a45fb496082c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1230,8 +1230,7 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) &soc_component_dev_aic32x4, &aic32x4_dai, 1); if (ret) { dev_err(dev, "Failed to register component\n"); - aic32x4_disable_regulators(aic32x4); - return ret; + goto err_disable_regulators; } if (gpio_is_valid(aic32x4->rstn_gpio)) { @@ -1242,9 +1241,14 @@ int aic32x4_probe(struct device *dev, struct regmap *regmap) ret = aic32x4_register_clocks(dev, aic32x4->mclk_name); if (ret) - return ret; + goto err_disable_regulators; return 0; + +err_disable_regulators: + aic32x4_disable_regulators(aic32x4); + + return ret; } EXPORT_SYMBOL(aic32x4_probe); -- 2.17.1

2 2

[PATCH] ASoC: topology: disable size checks for bytes_ext controls if needed
by Kai Vehmanen 17 Sep '20

17 Sep '20

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> When CONFIG_SND_CTL_VALIDATION is set, accesses to extended bytes control generate spurious error messages when the size exceeds 512 bytes, such as [ 11.224223] sof_sdw sof_sdw: control 2:0:0:EQIIR5.0 eqiir_coef_5:0: invalid count 1024 In addition the error check returns -EINVAL which has the nasty side effect of preventing applications accessing controls from working, e.g. root@plb:~# alsamixer cannot load mixer controls: Invalid argument It's agreed that the control interface has been abused since 2014, but forcing a check should not prevent existing solutions from working. This patch skips the checks conditionally if CONFIG_SND_CTL_VALIDATION is set and the byte array provided by topology is > 512. This preserves the checks for all other cases. Fixes: 1a3232d2f61d2 ('ASoC: topology: Add support for TLV bytes controls') BugLink: https://github.com/thesofproject/linux/issues/2430 Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(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/soc-topology.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 28faaf58326e..63086fa8b861 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -592,6 +592,17 @@ static int soc_tplg_kcontrol_bind_io(struct snd_soc_tplg_ctl_hdr *hdr, k->info = snd_soc_bytes_info_ext; k->tlv.c = snd_soc_bytes_tlv_callback; + /* + * When a topology-based implementation abuses the + * control interface and uses bytes_ext controls of + * more than 512 bytes, we need to disable the size + * checks, otherwise accesses to such controls will + * return an -EINVAL error and prevent the card from + * being configured. + */ + if (IS_ENABLED(CONFIG_SND_CTL_VALIDATION) && sbe->max > 512) + k->access |= SNDRV_CTL_ELEM_ACCESS_SKIP_CHECK; + ext_ops = tplg->bytes_ext_ops; num_ops = tplg->bytes_ext_ops_count; for (i = 0; i < num_ops; i++) { -- 2.27.0

2 1

[PATCH v5 0/5] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 17 Sep '20

17 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 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-cpu.c | 39 +- sound/soc/qcom/lpass-hdmi.c | 596 +++++++++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 129 +++++ 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 +++- 11 files changed, 1447 insertions(+), 154 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.

1 5

[PATCH] ASoC: rt715: Add module parameter to fix dmic pop sound issue.
by Pierre-Louis Bossart 17 Sep '20

17 Sep '20

From: Jack Yu <jack.yu(a)realtek.com> Add module parameter "power_up_delay" to fix pop noise on capture. The power_up_delay value is set with a default value of 400ms, smaller values are not recommended. BugLink: https://github.com/thesofproject/linux/issues/1969 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)linux.intel.com> Signed-off-by: Jack Yu <jack.yu(a)realtek.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/rt715.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index 099c8bd20006..0cf10dec1e3b 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -37,6 +37,10 @@ #include "rt715.h" +static int power_up_delay = 400; +module_param(power_up_delay, int, 0444); +MODULE_PARM_DESC(power_up_delay, "RT715 power up delay time in ms"); + static int rt715_index_write(struct regmap *regmap, unsigned int reg, unsigned int value) { @@ -498,6 +502,7 @@ static int rt715_set_bias_level(struct snd_soc_component *component, regmap_write(rt715->regmap, RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + msleep(power_up_delay); } break; -- 2.25.1

2 4

[PATCH 0/2] soundwire: sysfs: expose device number and status
by Pierre-Louis Bossart 17 Sep '20

17 Sep '20

This patchset combines three contributions: Srinivas Kandagalta suggested creating a device when 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 provide a nice addition for integrators/tests and were tested on Qualcomm and Intel platforms. Thanks Srinivas for the initial idea! 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 | 64 ++++++++++++++++++- 7 files changed, 109 insertions(+), 3 deletions(-) -- 2.25.1

2 5

[PATCH v2 0/2] soundwire: use {u32|u16}p_replace_bits
by Vinod Koul 17 Sep '20

17 Sep '20

This is v2 of the series to fix usage of FIELD_PREP and use {u32|u16}p_replace_bits. v1 used u32_replace_bits but looks like it might be removed so use the {u32|u16}p_replace_bits() instead. Also this helped to find that one instance in intel driver should have used u16p_replace_bits() so that is fixed too Vinod Koul (2): soundwire: cadence: use u32p_replace_bits soundwire: intel: use {u32|u16}p_replace_bits drivers/soundwire/cadence_master.c | 21 ++++++++++----------- drivers/soundwire/intel.c | 14 +++++++------- 2 files changed, 17 insertions(+), 18 deletions(-) -- 2.26.2

1 2

Re: [oss-drivers] [trivial PATCH] treewide: Convert switch/case fallthrough; to break;
by Simon Horman 17 Sep '20

17 Sep '20

On Wed, Sep 09, 2020 at 01:06:39PM -0700, Joe Perches wrote: > fallthrough to a separate case/default label break; isn't very readable. > > Convert pseudo-keyword fallthrough; statements to a simple break; when > the next label is case or default and the only statement in the next > label block is break; > > Found using: > > $ grep-2.5.4 -rP --include=*.[ch] -n "fallthrough;(\s*(case\s+\w+|default)\s*:\s*){1,7}break;" * > > Miscellanea: > > o Move or coalesce a couple label blocks above a default: block. > > Signed-off-by: Joe Perches <joe(a)perches.com> ... > diff --git a/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c b/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c > index 252fe06f58aa..1d5b87079104 100644 > --- a/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c > +++ b/drivers/net/ethernet/netronome/nfp/nfpcore/nfp6000_pcie.c > @@ -345,7 +345,7 @@ static int matching_bar(struct nfp_bar *bar, u32 tgt, u32 act, u32 tok, > baract = NFP_CPP_ACTION_RW; > if (act == 0) > act = NFP_CPP_ACTION_RW; > - fallthrough; > + break; > case NFP_PCIE_BAR_PCIE2CPP_MapType_FIXED: > break; > default: This is a cascading fall-through handling all map types. I don't think this change improves readability. ...

1 0

"aplay -D oss -l" still reads sound cards through ALSA, not OSS
by GitHub issues - edited 17 Sep '20

17 Sep '20

alsa-project/alsa-utils issue #56 was edited from yurivict: I don't have any ALSA devices, only OSS devices. When the OSS PCM is selected ```aplay``` still tries to read ALSA devices: ``` $ /usr/bin/aplay -D oss -l aplay: device_list:272: no soundcards found... ``` ``` $ /usr/bin/aplay -L null Discard all samples (playback) or generate zero samples (capture) oss Open Sound System ``` But ```/usr/bin/aplay x.wav``` plays it through the OSS card. aplay should list OSS cards when -D is selected. The ALSA interface should also list OSS cards when no ALSA cards are present but it doesn't. alsa-lib-1.1.8 alsa-plugins-oss-1.1.6 alsa-utils-1.1.8 Issue URL : https://github.com/alsa-project/alsa-utils/issues/56 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

"aplay -D oss -l" still reads sound cards through ALSA, not OSS
by GitHub issues - opened 17 Sep '20

17 Sep '20

alsa-project/alsa-utils issue #56 was opened from yurivict: I don't have any ALSA devices, only OSS devices. When the OSS PCM is selected ```aplay``` still tries to read ALSA devices: ``` $ /usr/bin/aplay -D oss -l aplay: device_list:272: no soundcards found... ``` ``` $ /usr/bin/aplay -L null Discard all samples (playback) or generate zero samples (capture) oss Open Sound System ``` alsa-lib-1.1.8 alsa-plugins-oss-1.1.6 alsa-utils-1.1.8 Issue URL : https://github.com/alsa-project/alsa-utils/issues/56 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/3] ASoC: fsl_sai: update the register list
by Shengjiu Wang 17 Sep '20

17 Sep '20

As sai ip is upgraded, so update sai register list. Shengjiu Wang (3): ASoC: fsl_sai: Add new added registers and new bit definition ASoC: fsl_sai: Add fsl_sai_check_version function ASoC: fsl_sai: Set MCLK input or output direction sound/soc/fsl/fsl_sai.c | 77 ++++++++++++++++++++++++++++++++++++ sound/soc/fsl/fsl_sai.h | 87 +++++++++++++++++++++++++++++++++++++++++ 2 files changed, 164 insertions(+) -- 2.27.0

3 9

Re: [PATCH] ASoC: fsl_audmix: make clock and output src write only
by Nicolin Chen 17 Sep '20

17 Sep '20

On Mon, Sep 14, 2020 at 08:24:34PM +0300, Viorel Suman (OSS) wrote: > From: Viorel Suman <viorel.suman(a)nxp.com> > > "alsactl -f state.conf store/restore" sequence fails because setting > "mixing clock source" and "output source" requires active TDM clock > being started for configuration propagation. Make these two controls > write only so that their values are not stored at "alsactl store". > > Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> > --- > sound/soc/fsl/fsl_audmix.c | 16 ++++++++++++---- > 1 file changed, 12 insertions(+), 4 deletions(-) > > diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c > index a447baf..7ad5925 100644 > --- a/sound/soc/fsl/fsl_audmix.c > +++ b/sound/soc/fsl/fsl_audmix.c > @@ -199,10 +199,18 @@ static int fsl_audmix_put_out_src(struct snd_kcontrol *kcontrol, > > static const struct snd_kcontrol_new fsl_audmix_snd_controls[] = { > /* FSL_AUDMIX_CTR controls */ > - SOC_ENUM_EXT("Mixing Clock Source", fsl_audmix_enum[0], > - snd_soc_get_enum_double, fsl_audmix_put_mix_clk_src), > - SOC_ENUM_EXT("Output Source", fsl_audmix_enum[1], > - snd_soc_get_enum_double, fsl_audmix_put_out_src), > + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "Mixing Clock Source", > + .info = snd_soc_info_enum_double, > + .access = SNDRV_CTL_ELEM_ACCESS_WRITE, > + .put = fsl_audmix_put_mix_clk_src, > + .private_value = (unsigned long)&fsl_audmix_enum[0] }, > + { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, > + .name = "Output Source", > + .info = snd_soc_info_enum_double, > + .access = SNDRV_CTL_ELEM_ACCESS_WRITE, > + .put = fsl_audmix_put_out_src, > + .private_value = (unsigned long)&fsl_audmix_enum[1] }, > SOC_ENUM("Output Width", fsl_audmix_enum[2]), > SOC_ENUM("Frame Rate Diff Error", fsl_audmix_enum[3]), > SOC_ENUM("Clock Freq Diff Error", fsl_audmix_enum[4]), > -- > 2.7.4 >

1 0

[PATCH v3 0/3] soundwire: qcom: fix IP version v1.5.1 support
by Srinivas Kandagatla 16 Sep '20

16 Sep '20

While testing Qualcomm soundwire controller version 1.5.1, found two issue, Firstly the frame shape information configured vs the bus parameters are out of sync. secondly some ports on this ip version require block packing mode support. With this patchset I was able to test 2 WSA speakers! Also I found a regression due to move to REG_FIELD, which patch 1 fixes it! thanks, srini Changes since v2: - updated to use u32_replace_bits(). - wrapped up in single lines as suggested by vkoul. Srinivas Kandagatla (3): soundwire: qcom: clear BIT FIELDs before value set. soundwire: qcom: add support to block packing mode soundwire: qcom: get max rows and cols info from compatible drivers/soundwire/qcom.c | 70 +++++++++++++++++++++++++++++----------- 1 file changed, 51 insertions(+), 19 deletions(-) -- 2.21.0

1 3

[PATCH v2 0/3] soundwire: qcom: fix IP version v1.5.1 support
by Srinivas Kandagatla 16 Sep '20

16 Sep '20

While testing Qualcomm soundwire controller version 1.5.1, found two issue, Firstly the frame shape information configured vs the bus parameters are out of sync. secondly some ports on this ip version require block packing mode support. With this patchset I was able to test 2 WSA speakers! Also I found a regression due to move to REG_FIELD, which patch 1 fixes it! thanks, srini Changes since v1: - rebased on top of REG_FILED patch or soundwire-next branch - udated qcom_swrm_data to use u32 instead of int as suggested by VKoul Srinivas Kandagatla (3): soundwire: qcom: clear BIT FIELDs before value set. soundwire: qcom: add support to block packing mode soundwire: qcom: get max rows and cols info from compatible drivers/soundwire/qcom.c | 76 ++++++++++++++++++++++++++++++---------- 1 file changed, 58 insertions(+), 18 deletions(-) -- 2.21.0

3 10

[PATCH 1/2] soundwire: cadence: use u32_replace_bits
by Vinod Koul 16 Sep '20

16 Sep '20

FIELD_PREP() does not replace the bits so it is not apt in case where we modify a register. Use u32_replace_bits() instead. Fixes: 3cf25d63b1b9 ("soundwire: cadence: use FIELD_{GET|PREP}") Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- drivers/soundwire/cadence_master.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index f1efe5beead6..bb293e7010f3 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -1278,9 +1278,9 @@ static int cdns_port_params(struct sdw_bus *bus, dpn_config = cdns_readl(cdns, dpn_config_off); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_WL, (p_params->bps - 1)); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_PORT_FLOW, p_params->flow_mode); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_PORT_DAT, p_params->data_mode); + dpn_config = u32_replace_bits(dpn_config, (p_params->bps - 1), CDNS_DPN_CONFIG_WL); + dpn_config = u32_replace_bits(dpn_config, p_params->flow_mode, CDNS_DPN_CONFIG_PORT_FLOW); + dpn_config = u32_replace_bits(dpn_config, p_params->data_mode, CDNS_DPN_CONFIG_PORT_DAT); cdns_writel(cdns, dpn_config_off, dpn_config); @@ -1316,18 +1316,19 @@ static int cdns_transport_params(struct sdw_bus *bus, } dpn_config = cdns_readl(cdns, dpn_config_off); - - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_BGC, t_params->blk_grp_ctrl); - dpn_config |= FIELD_PREP(CDNS_DPN_CONFIG_BPM, t_params->blk_pkg_mode); + dpn_config = u32_replace_bits(dpn_config, t_params->blk_grp_ctrl, CDNS_DPN_CONFIG_BGC); + dpn_config = u32_replace_bits(dpn_config, t_params->blk_pkg_mode, CDNS_DPN_CONFIG_BPM); cdns_writel(cdns, dpn_config_off, dpn_config); - dpn_offsetctrl |= FIELD_PREP(CDNS_DPN_OFFSET_CTRL_1, t_params->offset1); - dpn_offsetctrl |= FIELD_PREP(CDNS_DPN_OFFSET_CTRL_2, t_params->offset2); + dpn_offsetctrl = u32_replace_bits(dpn_offsetctrl, + t_params->offset1, CDNS_DPN_OFFSET_CTRL_1); + dpn_offsetctrl = u32_replace_bits(dpn_offsetctrl, + t_params->offset2, CDNS_DPN_OFFSET_CTRL_2); cdns_writel(cdns, dpn_offsetctrl_off, dpn_offsetctrl); - dpn_hctrl |= FIELD_PREP(CDNS_DPN_HCTRL_HSTART, t_params->hstart); - dpn_hctrl |= FIELD_PREP(CDNS_DPN_HCTRL_HSTOP, t_params->hstop); - dpn_hctrl |= FIELD_PREP(CDNS_DPN_HCTRL_LCTRL, t_params->lane_ctrl); + dpn_hctrl = u32_replace_bits(dpn_hctrl, t_params->hstart, CDNS_DPN_HCTRL_HSTART); + dpn_hctrl = u32_replace_bits(dpn_hctrl, t_params->hstop, CDNS_DPN_HCTRL_HSTOP); + dpn_hctrl = u32_replace_bits(dpn_hctrl, t_params->lane_ctrl, CDNS_DPN_HCTRL_LCTRL); cdns_writel(cdns, dpn_hctrl_off, dpn_hctrl); cdns_writel(cdns, dpn_samplectrl_off, (t_params->sample_interval - 1)); -- 2.26.2

1 1

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel without altering the volume level of right channel as we can expect. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel without altering the volume level as we can expect. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2 0/3] regmap: add SoundWire 1.2 MBQ support
by Pierre-Louis Bossart 16 Sep '20

16 Sep '20

In preparation of the upstream contribution of SDCA (SoundWire Device Class for Audio) ASoC codec drivers [1] [2], add regmap support SoundWire 1.2 MBQ support. The MBQ (Multi-Byte Quantity) registers need to be handled in a different way from regular 8-bit SoundWire registers, their main application is going to be for volume/gain controls. The second patch was initially suggested for inclusion in the SoundWire tree, and was modified to add more background information on SDCA in the commit message as requested by Vinod Koul. Changes since v1: Rebased on regmap tree (conflict with SPI stuff). Removed mod_devicetable.h header Removed -EOPNOTSUPP error codes, use -ENOTSUPP Added long description of SDCA Used FIELD_PREP/GET as suggested by Vinod Koul Added Bard Liao's Acked-by tag. Pierre-Louis Bossart (3): regmap: sdw: add required header files soundwire: SDCA: add helper macro to access controls regmap: sdw: add support for SoundWire 1.2 MBQ drivers/base/regmap/Kconfig | 6 +- drivers/base/regmap/Makefile | 1 + drivers/base/regmap/regmap-sdw-mbq.c | 101 ++++++++++++++++++++++++ drivers/base/regmap/regmap-sdw.c | 2 + include/linux/regmap.h | 21 +++++ include/linux/soundwire/sdw_registers.h | 33 ++++++++ 6 files changed, 163 insertions(+), 1 deletion(-) create mode 100644 drivers/base/regmap/regmap-sdw-mbq.c base-commit: d17343b87da422a59d99a3ed130573dbeb96c582 -- 2.25.1

2 17

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel as we can expect. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - edited 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was edited from maxmbed: ### Description I am encountering an unexpected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel as we can expected. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aplay: playing mono audio file alteres pulseaudio default Master left or right volume level
by GitHub issues - opened 16 Sep '20

16 Sep '20

alsa-project/alsa-utils issue #55 was opened from maxmbed: ### Description I am encountering an expected behaviour with pulse audio master volume in my application. If I set for example left channel to 100% and right channel 0%, and play mono audio file (i.e. audio file with 1 channel only), then I would expect to hear only the audio file on left channel. The actual behaviour in the example of above is when playing the audio file, the Master volume right channel will be automatically adjusted to 50%. If play again it increases to 75% and so on. This issue happens also with other R/L volume level, the case of above is just an example. **Important to note the case does not happen with stereo audio file** where audio sounds on left only channel as we can expected. ### Steps to repeat issue: (a) Set mixer Lch to 100%, Rch to 0% ``` $ amixer -D default sset Master 65535,0 Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 65536 Mono: Front Left: Playback 65535 [100%] [on] Front Right: Playback 0 [0%] [on] ``` (b) Play mono audio file ``` $ aplay -D default ringtone.wav Playing WAVE 'ringtone.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono ``` ### Request (1) Is it by design expected ? (2) Is there a way to prevent this phenomenon (command option) ? Issue URL : https://github.com/alsa-project/alsa-utils/issues/55 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Re: [PATCH] fsl: imx-audmix: Use devm_kcalloc() in imx_audmix_probe()
by Markus Elfring 16 Sep '20

16 Sep '20

> A multiplication for the size determination of a memory allocation > indicated that an array data structure should be processed. > Thus use the corresponding function "devm_kcalloc". I find the previous patch subject inappropriate. Would you like to choose an alternative? Regards, Markus

1 0

Re: [PATCH v2 3/3] ASoC: tlv320adcx140: Add proper support for master mode
by Dan Murphy 16 Sep '20

16 Sep '20

Camel On 9/16/20 2:52 AM, Camel Guo wrote: > Please forget about this patch since Dan will upload a similar one. > > @Dan, see my comment below. > I have cc'd you on my patchset and as you can see I did not add any of the master mode programming only setting of the master mode bits. So this patch is still viable it's just the master mode bit programming that needs to be looked at. Dan

1 0

[PATCH v3 00/11] USB: new USB control message helper functions
by Greg Kroah-Hartman 16 Sep '20

16 Sep '20

In a recent discussion about a USB networking bug found by syzbot, and fixed by Himadri Pandya, the design of the existing usb_control_msg() call was brought up as not being the "nicest" thing to use by Dmitry Vyukov: https://lore.kernel.org/r/CACT4Y+YbDODLRFn8M5QcY4CazhpeCaunJnP_udXtAs0rYoAS… The function makes it hard to get right, in that it will return the number of bytes sent/received, but almost no one checks to see if a short read/write happens. With a malicious, or broken, USB device, this can cause drivers to act on data that they did not anticipate, and sometimes copy internal kernel data out to userspace. So let's fix this up by creating two new functions, usb_control_msg_send() and usb_control_msg_recv(). These functions either complete the full transation, or they return an error, a short send/recv is now an error. They also accept data off of the stack, saving individual drivers the pain of having to constantly allocate memory on their own for tiny messages, thereby saving overall kernel code space. The api also does not require a raw USB "pipe" to be sent to the function, as we know the direction, so just pass in the endpoint number instead, again making it easier on the USB driver author to use. This series first takes a helper function out of the sound core for verifying USB endpoints to be able to use internally, and then adds the new functions, converts over some internal USB code to use them, and then starts to clean up some drivers using these new functions, as an example of the savings that can happen by using these functions. Thanks to Dmitry and Himadri for the idea on how to do all of this. greg k-h ----- Changes from v2: - add Andy's reviewed-by: to patch 3 - remove unneeded change in usb_enable_link_state() in hub.c in patch 4 Changes from v1: - added acks from Takashi Iwai - dropped changes to one function in patch 04 thanks to review from Alan Stern - typo fix in comment in patch 01 - added new patch 11 to remove some unneeded checks in the sound drivers for endpoint statuses that would always be true. Greg Kroah-Hartman (11): USB: move snd_usb_pipe_sanity_check into the USB core USB: add usb_control_msg_send() and usb_control_msg_recv() USB: core: message.c: use usb_control_msg_send() in a few places USB: core: hub.c: use usb_control_msg_send() in a few places USB: legousbtower: use usb_control_msg_recv() sound: usx2y: move to use usb_control_msg_send() sound: 6fire: move to use usb_control_msg_send() and usb_control_msg_recv() sound: line6: move to use usb_control_msg_send() and usb_control_msg_recv() sound: hiface: move to use usb_control_msg_send() Bluetooth: ath3k: use usb_control_msg_send() and usb_control_msg_recv() ALSA: remove calls to usb_pipe_type_check for control endpoints drivers/bluetooth/ath3k.c | 90 +++++------------ drivers/usb/core/hub.c | 99 ++++++++---------- drivers/usb/core/message.c | 171 ++++++++++++++++++++++++++++---- drivers/usb/core/urb.c | 31 ++++-- drivers/usb/misc/legousbtower.c | 60 ++++------- include/linux/usb.h | 7 ++ sound/usb/6fire/firmware.c | 38 +++---- sound/usb/helper.c | 16 +-- sound/usb/helper.h | 1 - sound/usb/hiface/pcm.c | 14 ++- sound/usb/line6/driver.c | 69 +++++-------- sound/usb/line6/podhd.c | 17 ++-- sound/usb/line6/toneport.c | 8 +- sound/usb/mixer_scarlett_gen2.c | 2 +- sound/usb/quirks.c | 12 +-- sound/usb/usx2y/us122l.c | 42 ++------ 16 files changed, 336 insertions(+), 341 deletions(-) -- 2.28.0

2 13

[PATCH 01/10] dt-bindings: arm: samsung: pmu: Use unevaluatedProperties
by Krzysztof Kozlowski 16 Sep '20

16 Sep '20

Additional properties actually might appear (e.g. assigned-clocks) so use unevaluatedProperties to fix dtbs_check warnings like: arch/arm64/boot/dts/exynos/exynos5433-tm2.dt.yaml: system-controller@105c0000: 'assigned-clock-parents', 'assigned-clocks' do not match any of the regexes: 'pinctrl-[0-9]+' Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Documentation/devicetree/bindings/arm/samsung/pmu.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/arm/samsung/pmu.yaml b/Documentation/devicetree/bindings/arm/samsung/pmu.yaml index 686c13c14e32..30ff2da81416 100644 --- a/Documentation/devicetree/bindings/arm/samsung/pmu.yaml +++ b/Documentation/devicetree/bindings/arm/samsung/pmu.yaml @@ -86,7 +86,7 @@ required: - compatible - reg -additionalProperties: false +unevaluatedProperties: false allOf: - if: -- 2.17.1

6 35

[PATCH v2 1/3] dt-bindings: tlv320adcx140: Add GPIO config and drive config
by Camel Guo 15 Sep '20

15 Sep '20

From: Camel Guo <camelg(a)axis.com> Add properties for configuring the General Purpose Input Outputs (GPIO). There are 2 settings for GPIO, configuration and the output drive type. Signed-off-by: Camel Guo <camelg(a)axis.com> --- .../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..7b0b4554da59 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 driver 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

2 6

Re: [PATCH v4 2/5] ASoC: dt-bindings: Add dt binding for lpass hdmi
by Rob Herring 15 Sep '20

15 Sep '20

On Tue, Sep 08, 2020 at 05:44:39PM +0000, Srinivasa Rao Mandadapu wrote: > From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> > > Adds bindings for lpass hdmi interface > which can support audio path over dp. > > Signed-off-by: Srinivasa Rao <srivasam(a)codeaurora.org> > Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> > --- > .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 51 ++++++++++++++++++++-- > 1 file changed, 47 insertions(+), 4 deletions(-) > > diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml > index 09c9bd2..7c2ac0c 100644 > --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml > +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml > @@ -22,6 +22,7 @@ properties: > - qcom,lpass-cpu > - qcom,apq8016-lpass-cpu > - qcom,sc7180-lpass-cpu > + - qcom,sc7180-lpass-hdmi > > reg: > maxItems: 1 > @@ -60,10 +61,12 @@ properties: > const: 0 > > patternProperties: > - "(^mi2s-[0-9a-f]$|mi2s)": > + "^dai@[0-9a-f]$": > type: object > - description: Required properties for each DAI > - > + description: | > + LPASS CPU dai node for each I2S device. Bindings of each node > + depends on the specific driver providing the functionality and > + properties. > properties: > reg: > maxItems: 1 > @@ -145,6 +148,22 @@ allOf: > - iommus > - power-domains > > + - if: > + properties: > + compatible: > + contains: > + const: qcom,sc7180-lpass-hdmi > + then: > + properties: > + clock-names: > + items: > + - const: pcnoc-sway-clk > + - const: audio-core > + - const: pcnoc-mport-clk > + required: > + - iommus > + - power-domains > + > examples: > - | > #include <dt-bindings/sound/sc7180-lpass.h> > @@ -178,12 +197,36 @@ examples: > #address-cells = <1>; > #size-cells = <0>; > /* Optional to set different MI2S SD lines */ > - mi2s-primary@0 { > + dai@mi2s-primary { The unit address should be a number. As this is not failing checks, then you are missing an 'additionalProperties: false'. > reg = <MI2S_PRIMARY>; > qcom,playback-sd-lines = <1>; > qcom,capture-sd-lines = <0>; > }; > }; > + > + lpassh@62d87000 { > + compatible = "qcom,sc7180-lpass-hdmi"; > + > + reg = <0 0x62d87000 0 0x68000>; > + > + iommus = <&apps_smmu 0x1032 0>; > + > + power-domains = <&lpass_hm 0>; > + > + clocks = <&gcc 131>, > + <&lpasscc 6>, > + <&lpasscc 10>; > + > + clock-names = "pcnoc-sway-clk", "audio-core", > + "pcnoc-mport-clk"; > + > + #sound-dai-cells = <1>; > + > + #address-cells = <1>; > + #size-cells = <0>; > + > + interrupts = <0 268 1>; > + }; > }; > > ... > -- > Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., > is a member of Code Aurora Forum, a Linux Foundation Collaborative Project. >

1 0

[RFC] dpcm_fe_dai_do_trigger: implicit removal of _DRAIN cmd since v5.4
by Cezary Rojewski 15 Sep '20

15 Sep '20

Hello, Till v5.4 dpcm_fe_dai_do_trigger was supporting all pcm cmds as each case: SND_SOC_DPCM_TRIGGER_PRE/ _POST/ _BESPOKE was simply passing cmd's value to appropriate variant of _trigger(). Since the addition of: acbf27746ecfa96b290b54cc7f05273482ea128a ASoC: pcm: update FE/BE trigger order based on the command additional filters have been introduced for _PRE and _POST cases: switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = dpcm_dai_trigger_fe_be(substream, cmd, true); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: ret = dpcm_dai_trigger_fe_be(substream, cmd, false); break; default: ret = -EINVAL; break; } effectively removing support for _DRAIN command - all requests will yield -EINVAL. _BESPOKE was left alone so support remained there. Now, is DPCM no longer supporting _DRAIN and that's how things should be -or- DPCM still intends to support _DRAIN and mentioned change is unintended regression? Czarek

2 1

Re: [PATCH v4 1/5] ASoC: Add sc7180-lpass binding header hdmi define
by Rob Herring 15 Sep '20

15 Sep '20

On Tue, 08 Sep 2020 17:44:33 +0000, Srinivasa Rao Mandadapu wrote: > From: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> > > Add header defining hdmi dai-id for SC7180 lpass soc > in dt bindings. > > Signed-off-by: Srinivasa Rao <srivasam(a)codeaurora.org> > Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> > --- > include/dt-bindings/sound/sc7180-lpass.h | 1 + > 1 file changed, 1 insertion(+) > Acked-by: Rob Herring <robh(a)kernel.org>

1 0

a saner API for allocating DMA addressable pages v2
by Christoph Hellwig 15 Sep '20

15 Sep '20

Hi all, this series replaced the DMA_ATTR_NON_CONSISTENT flag to dma_alloc_attrs with a separate new dma_alloc_pages API, which is available on all platforms. In addition to cleaning up the convoluted code path, this ensures that other drivers that have asked for better support for non-coherent DMA to pages with incurring bounce buffering over can finally be properly supported. I'm still a little unsure about the API naming, as alloc_pages sort of implies a struct page return value, but we return a kernel virtual address. The other alternative would be to name the API dma_alloc_noncoherent, but the whole non-coherent naming seems to put people off. As a follow up I plan to move the implementation of the DMA_ATTR_NO_KERNEL_MAPPING flag over to this framework as well, given that is also is a fundamentally non coherent allocation. The replacement for that flag would then return a struct page, as it is allowed to actually return pages without a kernel mapping as the name suggested (although most of the time they will actually have a kernel mapping..) In addition to the conversions of the existing non-coherent DMA users, I've also added a patch to convert the firewire ohci driver to use the new dma_alloc_pages API. Note that I haven't carried over any Tested-by: tags for the noncoherent allocation conversions as there was a bit of a patch reshuffle, but the result should be the same. The first patch is queued up for 5.9 in the media tree, but included here for completeness. A git tree is available here: git://git.infradead.org/users/hch/misc.git dma_alloc_pages Gitweb: http://git.infradead.org/users/hch/misc.git/shortlog/refs/heads/dma_alloc_p… Changes since v1: - rebased on the latests dma-mapping tree, which merged many of the cleanups - fix an argument passing typo in 53c700, caught by sparse - rename a few macro arguments in 53c700 - pass the right device to the DMA API in the lib82596 drivers - fix memory ownershiptransfers in sgiseeq - better document what a page in the direct kernel mapping means - split into dma_alloc_pages that returns a struct page and is in the direct mapping vs dma_alloc_noncoherent that can be vmapped - conver the firewire ohci driver to dma_alloc_pages Diffstat:

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Re: [PATCH 03/17] drm/exynos: stop setting DMA_ATTR_NON_CONSISTENT
by Christoph Hellwig 15 Sep '20

15 Sep '20

On Mon, Sep 14, 2020 at 06:34:02PM +0300, Sergei Shtylyov wrote: > On 9/14/20 5:44 PM, Christoph Hellwig wrote: > > > DMA_ATTR_NON_CONSISTENT is a no-op except on PARISC and some mips > > configs, so don't set it in this ARM specific driver. > > Hm, PARICS and ARM capitalized but mips in lower case? :-) I guess it should also be PA-RISC if we start nitpicking..

1 0

Possible bug for ubuntu 20.04 with CONFIG_SND_MAX_CARDS=256
by Olivier SCHMITT 15 Sep '20

15 Sep '20

Hello dear all, I have been compiling kernels for a long time with CONFIG_SND_MAX_CARDS = 256 and increasing the sample rate. I switched to ubuntu 20.04 server lts. Everything is working fine until I compile alsa-lib. git clone git: //git.alsa-project.org/alsa-lib.git alsa-lib ./gitcompile --with-max-cards = 256 But after the make install, the plughw, dmix and dnsoop disappear. Do you have an idea? I uploaded alsa-info on: http://alsa-project.org/db/?f=8860b8bbb5abd5a84d3002717d68206db7da4a9c Thanks in advance.

1 0

[PATCH 1/2] leds: trigger: audio: Add audio jack plugging indicator led
by Hui Wang 15 Sep '20

15 Sep '20

On the Lenovo P520 front panel, there is a headset plugging indicator led. According to LENOVO's requirement, this led should be on if the front headset audio jack is plugged, otherwise the led should be off. We tested with windows, if we plugged a headphone/headset in that jack, the led was on, if we removed the headphone/headset from that jack, the led was off. Here we add JACKPLUG led in the ledtrig_audio, then we could call ledtrig_audio_set() in the alsa driver to control that led. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- drivers/leds/trigger/Kconfig | 4 ++-- drivers/leds/trigger/ledtrig-audio.c | 7 +++++-- include/linux/leds.h | 1 + 3 files changed, 8 insertions(+), 4 deletions(-) diff --git a/drivers/leds/trigger/Kconfig b/drivers/leds/trigger/Kconfig index ce9429ca6dde..a61893a6e3e4 100644 --- a/drivers/leds/trigger/Kconfig +++ b/drivers/leds/trigger/Kconfig @@ -138,10 +138,10 @@ config LEDS_TRIGGER_PATTERN If unsure, say N config LEDS_TRIGGER_AUDIO - tristate "Audio Mute LED Trigger" + tristate "Audio Mute and Jack Plugging LED Trigger" help This allows LEDs to be controlled by audio drivers for following - the audio mute and mic-mute changes. + the audio mute, mic-mute and jack plugging changes. If unsure, say N endif # LEDS_TRIGGERS diff --git a/drivers/leds/trigger/ledtrig-audio.c b/drivers/leds/trigger/ledtrig-audio.c index f76621e88482..fa66bb982400 100644 --- a/drivers/leds/trigger/ledtrig-audio.c +++ b/drivers/leds/trigger/ledtrig-audio.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// Audio Mute LED trigger +// Audio Mute And Jack Plugging LED trigger // #include <linux/kernel.h> @@ -29,6 +29,8 @@ static int __init ledtrig_audio_init(void) &ledtrig_audio[LED_AUDIO_MUTE]); led_trigger_register_simple("audio-micmute", &ledtrig_audio[LED_AUDIO_MICMUTE]); + led_trigger_register_simple("audio-jackplug", + &ledtrig_audio[LED_AUDIO_JACKPLUG]); return 0; } module_init(ledtrig_audio_init); @@ -37,8 +39,9 @@ static void __exit ledtrig_audio_exit(void) { led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_MUTE]); led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_MICMUTE]); + led_trigger_unregister_simple(ledtrig_audio[LED_AUDIO_JACKPLUG]); } module_exit(ledtrig_audio_exit); -MODULE_DESCRIPTION("LED trigger for audio mute control"); +MODULE_DESCRIPTION("LED trigger for audio mute and jack plugging control"); MODULE_LICENSE("GPL v2"); diff --git a/include/linux/leds.h b/include/linux/leds.h index 6a8d6409c993..1858c2d30ecd 100644 --- a/include/linux/leds.h +++ b/include/linux/leds.h @@ -584,6 +584,7 @@ struct led_pattern { enum led_audio { LED_AUDIO_MUTE, /* master mute LED */ LED_AUDIO_MICMUTE, /* mic mute LED */ + LED_AUDIO_JACKPLUG, /* audio jack plug indicator LED */ NUM_AUDIO_LEDS }; -- 2.17.1

1 2

increase CONFIG_SND_MAX_CARDS and avalable sample rate
by Olivier SCHMITT 14 Sep '20

14 Sep '20

Hello dear all, It's possible de increase CONFIG_SND_MAX_CARDS to 256 and increse sample rate to 384khz. Actualy i build my own kernel as: sed -i 's/SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_192000,/SNDRV_PCM_RATE_CONTINUOUS,/g' ./sound/drivers/aloop.c sed -i 's/192000/768000/g' ./sound/drivers/aloop.c grep -rnl './' -e "CONFIG_SND_MAX_CARDS=32" | xargs -i@ sed -i 's/CONFIG_SND_MAX_CARDS=32/CONFIG_SND_MAX_CARDS=256/g' @ search and replace 32 by 256 in and ./debian.master/config/annotations:CONFIG_SND_MAX_CARDS and for alsa utils: git clone git://git.alsa-project.org/alsa-lib.git alsa-lib ./gitcompile --with-max-cards=256 make install git clone git://git.alsa-project.org/alsa-utils.git alsa-utils ./gitcompile --with-max-cards=256 make install Thanks in advance.

1 0

[PATCH v2 00/11] USB: new USB control message helper functions
by Greg Kroah-Hartman 14 Sep '20

14 Sep '20

In a recent discussion about a USB networking bug found by syzbot, and fixed by Himadri Pandya, the design of the existing usb_control_msg() call was brought up as not being the "nicest" thing to use by Dmitry Vyukov: https://lore.kernel.org/r/CACT4Y+YbDODLRFn8M5QcY4CazhpeCaunJnP_udXtAs0rYoAS… The function makes it hard to get right, in that it will return the number of bytes sent/received, but almost no one checks to see if a short read/write happens. With a malicious, or broken, USB device, this can cause drivers to act on data that they did not anticipate, and sometimes copy internal kernel data out to userspace. So let's fix this up by creating two new functions, usb_control_msg_send() and usb_control_msg_recv(). These functions either complete the full transation, or they return an error, a short send/recv is now an error. They also accept data off of the stack, saving individual drivers the pain of having to constantly allocate memory on their own for tiny messages, thereby saving overall kernel code space. The api also does not require a raw USB "pipe" to be sent to the function, as we know the direction, so just pass in the endpoint number instead, again making it easier on the USB driver author to use. This series first takes a helper function out of the sound core for verifying USB endpoints to be able to use internally, and then adds the new functions, converts over some internal USB code to use them, and then starts to clean up some drivers using these new functions, as an example of the savings that can happen by using these functions. Thanks to Dmitry and Himadri for the idea on how to do all of this. greg k-h Changes from v1: - added acks from Takashi Iwai - dropped changes to one function in patch 04 thanks to review from Alan Stern - typo fix in comment in patch 01 - added new patch 11 to remove some unneeded checks in the sound drivers for endpoint statuses that would always be true. *** BLURB HERE *** Greg Kroah-Hartman (11): USB: move snd_usb_pipe_sanity_check into the USB core USB: add usb_control_msg_send() and usb_control_msg_recv() USB: core: message.c: use usb_control_msg_send() in a few places USB: core: hub.c: use usb_control_msg_send() in a few places USB: legousbtower: use usb_control_msg_recv() sound: usx2y: move to use usb_control_msg_send() sound: 6fire: move to use usb_control_msg_send() and usb_control_msg_recv() sound: line6: move to use usb_control_msg_send() and usb_control_msg_recv() sound: hiface: move to use usb_control_msg_send() Bluetooth: ath3k: use usb_control_msg_send() and usb_control_msg_recv() ALSA: remove calls to usb_pipe_type_check for control endpoints drivers/bluetooth/ath3k.c | 90 +++++------------ drivers/usb/core/hub.c | 99 ++++++++---------- drivers/usb/core/message.c | 171 ++++++++++++++++++++++++++++---- drivers/usb/core/urb.c | 31 ++++-- drivers/usb/misc/legousbtower.c | 60 ++++------- include/linux/usb.h | 7 ++ sound/usb/6fire/firmware.c | 38 +++---- sound/usb/helper.c | 16 +-- sound/usb/helper.h | 1 - sound/usb/hiface/pcm.c | 14 ++- sound/usb/line6/driver.c | 69 +++++-------- sound/usb/line6/podhd.c | 17 ++-- sound/usb/line6/toneport.c | 8 +- sound/usb/mixer_scarlett_gen2.c | 2 +- sound/usb/quirks.c | 12 +-- sound/usb/usx2y/us122l.c | 42 ++------ 16 files changed, 335 insertions(+), 342 deletions(-) -- 2.28.0

3 15

Re: [PATCH] ASoC: tas6424: disable regulators when probe error
by Mark Brown 14 Sep '20

14 Sep '20

On Mon, Sep 14, 2020 at 11:52:15PM +0900, Chanho Park wrote: > On Mon, Sep 14, 2020 at 8:27 PM Mark Brown <broonie(a)kernel.org> wrote: > > Please think hard before including complete backtraces in upstream > > reports, they are very large and contain almost no useful information > > relative to their size so often obscure the relevant content in your > > message. If part of the backtrace is usefully illustrative (it often is > > for search engines if nothing else) then it's usually better to pull out > > the relevant sections. > Thanks for your suggestion. > I'll respin this patch without the call stack messages. It's fine, I've already applied it - just please bear this in mind for future things.

1 0

[PATCH] ALSA: rockchip_i2s: fix a possible divide-by-zero bug in rockchip_i2s_hw_params()
by Tuo Li 14 Sep '20

14 Sep '20

The variable bclk_rate is checked in: if (bclk_rate && mclk_rate % bclk_rate) This indicates that bclk_rate can be zero. If so, a divide-by-zero bug will occur: div_bclk = mclk_rate / bclk_rate; To fix this possible bug, the function returns -EINVAL when bclk_rate is zero. Signed-off-by: Tuo Li <tuoli96(a)outlook.com> --- sound/soc/rockchip/rockchip_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index d1438753edb4..593299675b8c 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -279,7 +279,7 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, if (i2s->is_master_mode) { mclk_rate = clk_get_rate(i2s->mclk); bclk_rate = 2 * 32 * params_rate(params); - if (bclk_rate && mclk_rate % bclk_rate) + if (bclk_rate == 0 || mclk_rate % bclk_rate) return -EINVAL; div_bclk = mclk_rate / bclk_rate; -- 2.17.1

2 1

[PATCH 0/4] ASoC: SOF: DSP core management fixes for 5.10
by Kai Vehmanen 14 Sep '20

14 Sep '20

This series contains some improvements to how DSP core management is done in SOF, and adds a distinction between cores managed by the host versus cores managed by the DSP. Pierre-Louis Bossart (1): ASoC: SOF: Intel: hda-loader: s/master/primary Ranjani Sridharan (3): ASoC: SOF: rename cores_mask to host_managed_cores_mask ASoC: SOF: Intel: hda: modify core_power_up/down op ASoC: SOF: Intel: remove the HDA_DSP_CORE_MASK() macro sound/soc/sof/intel/apl.c | 2 +- sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 6 +++--- sound/soc/sof/intel/cnl.c | 15 ++++----------- sound/soc/sof/intel/hda-dsp.c | 20 +++++++++++++++++--- sound/soc/sof/intel/hda-loader.c | 11 +++++------ sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/hda.h | 3 --- sound/soc/sof/intel/shim.h | 2 +- sound/soc/sof/intel/tgl.c | 2 +- 10 files changed, 34 insertions(+), 31 deletions(-) -- 2.27.0

2 5

[PATCH] ASoC: SOF: Intel: Use DMI oem string search for tgl_max98373_rt5682
by Kai Vehmanen 14 Sep '20

14 Sep '20

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> DMI product name is used to support system variants based out of tgl_max98373_rt5682 in current implementation. Replace this DMI search with DMI_OEM_STRING. Coreboot(BIOS used in these systems) is setting the needed DMI_OEM_STRING field to uniquely identify these systems. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.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> --- sound/soc/intel/boards/sof_rt5682.c | 2 +- sound/soc/sof/sof-pci-dev.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 9a6f10ede427..ddbb9fe7cc06 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -123,7 +123,7 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { .callback = sof_rt5682_quirk_cb, .matches = { DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Volteer"), - DMI_MATCH(DMI_PRODUCT_NAME, "Terrador"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98373_ALC5682I_I2S_UP4"), }, .driver_data = (void *)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index ec62d9337d68..3ec380945466 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -50,7 +50,7 @@ static const struct dmi_system_id sof_tplg_table[] = { .callback = sof_tplg_cb, .matches = { DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Volteer"), - DMI_MATCH(DMI_PRODUCT_NAME, "Terrador"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98373_ALC5682I_I2S_UP4"), }, .driver_data = "sof-tgl-rt5682-ssp0-max98373-ssp2.tplg", }, -- 2.27.0

2 1

[PATCH 0/2] ASoC: mediatek: mt8183-da7219: support machine driver for rt1015p
by Tzung-Bi Shih 14 Sep '20

14 Sep '20

The series reuses mt8183-da7219-max98357.c for supporting machine driver with rt1015p speaker amplifier. The 1st patch adds document for the new proposed compatible string. The 2nd patch changes the machine driver to support "RT1015P" codec. Tzung-Bi Shih (2): ASoC: dt-bindings: mt8183-da7219: add compatible string for using rt1015p ASoC: mediatek: mt8183-da7219: support machine driver with rt1015p .../bindings/sound/mt8183-da7219-max98357.txt | 1 + sound/soc/mediatek/Kconfig | 1 + .../mediatek/mt8183/mt8183-da7219-max98357.c | 40 +++++++++++++++++++ 3 files changed, 42 insertions(+) -- 2.28.0.526.ge36021eeef-goog

2 3

[PATCH 1/2] ASoC: rt1015: Fix DC calibration on bypass boost mode
by derek.fang＠realtek.com 14 Sep '20

14 Sep '20

From: "derek.fang" <derek.fang(a)realtek.com> Fix the DC calibration unsuccessful issue on rt1015 bypass boost mode. Signed-off-by: derek.fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt1015.c | 48 +++++++++++++++++++++++++++++++++-------------- sound/soc/codecs/rt1015.h | 1 + 2 files changed, 35 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 548f686..ba1b6b3 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -484,6 +484,33 @@ static int rt1015_bypass_boost_get(struct snd_kcontrol *kcontrol, return 0; } +static void rt1015_calibrate(struct rt1015_priv *rt1015) +{ + struct snd_soc_component *component = rt1015->component; + struct regmap *regmap = rt1015->regmap; + + snd_soc_dapm_mutex_lock(&component->dapm); + regcache_cache_bypass(regmap, true); + + regmap_write(regmap, RT1015_PWR1, 0xd7df); + regmap_write(regmap, RT1015_PWR4, 0x00b2); + regmap_write(regmap, RT1015_CLSD_INTERNAL8, 0x2008); + regmap_write(regmap, RT1015_CLSD_INTERNAL9, 0x0140); + regmap_write(regmap, RT1015_GAT_BOOST, 0x0efe); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000d); + regmap_write(regmap, RT1015_PWR_STATE_CTRL, 0x000e); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a00); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a01); + regmap_write(regmap, RT1015_DC_CALIB_CLSD1, 0x5a05); + msleep(500); + regmap_write(regmap, RT1015_PWR1, 0x0); + + regcache_cache_bypass(regmap, false); + regcache_mark_dirty(regmap); + regcache_sync(regmap); + snd_soc_dapm_mutex_unlock(&component->dapm); +} + static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -494,20 +521,12 @@ static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, if (!rt1015->dac_is_used) { rt1015->bypass_boost = ucontrol->value.integer.value[0]; - if (rt1015->bypass_boost == RT1015_Bypass_Boost) { - snd_soc_component_write(component, - RT1015_PWR4, 0x00b2); - snd_soc_component_write(component, - RT1015_CLSD_INTERNAL8, 0x2008); - snd_soc_component_write(component, - RT1015_CLSD_INTERNAL9, 0x0140); - snd_soc_component_write(component, - RT1015_GAT_BOOST, 0x0efe); - snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x000d); - msleep(500); - snd_soc_component_write(component, - RT1015_PWR_STATE_CTRL, 0x000e); + if (rt1015->bypass_boost == RT1015_Bypass_Boost && + !rt1015->cali_done) { + rt1015_calibrate(rt1015); + rt1015->cali_done = 1; + + regmap_write(rt1015->regmap, RT1015_MONO_DYNA_CTRL, 0x0010); } } else dev_err(component->dev, "DAC is being used!\n"); @@ -888,6 +907,7 @@ static int rt1015_probe(struct snd_soc_component *component) rt1015->component = component; rt1015->bclk_ratio = 0; + rt1015->cali_done = 0; snd_soc_component_write(component, RT1015_BAT_RPO_STEP1, 0x061c); return 0; diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 7bd159e..4d11f58 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -389,6 +389,7 @@ struct rt1015_priv { int bypass_boost; int amp_ver; int dac_is_used; + int cali_done; }; #endif /* __RT1015_H__ */ -- 2.7.4

2 2

[PATCH v2 0/2] ASoC: ti: j721e-evm: Support for j7200 variant
by Peter Ujfalusi 14 Sep '20

14 Sep '20

Hi, Changes since v1: - Suffix the 2359296000 constant with 'u' to silence C90 warning When j7200 SOM is connected to the CPB, the audio setup is a bit different: Only 48KHz family have clock path, 44.1KHz is not supported. Update the binding documentation and add support for the j7200 version of CPB to the driver. Regards, Peter --- Peter Ujfalusi (2): ASoC: dt-bindings: ti,j721e-cpb-audio: Document support for j7200-cpb ASoC: ti: j721e-evm: Add support for j7200-cpb audio .../bindings/sound/ti,j721e-cpb-audio.yaml | 92 ++++++++++++++----- sound/soc/ti/j721e-evm.c | 11 +++ 2 files changed, 81 insertions(+), 22 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 3

[PATCH] ASoC: stm32: sai: add pm_runtime support
by Olivier Moysan 14 Sep '20

14 Sep '20

Enable support of pm_runtime on STM32 SAI driver to allow SAI power state monitoring. pm_runtime_put_autosuspend() is called from ASoC framework on pcm device close. The pmdown_time delay is available in runtime context, and may be set in SAI driver to take into account shutdown delay on playback. However, this shutdown delay is already handled in the DAPMs of the audio codec linked to SAI CPU DAI. So, the choice is made, not to support this delay on CPU DAI side. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3fb9513cedb2..3aa1cf262402 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/of_irq.h> #include <linux/of_platform.h> +#include <linux/pm_runtime.h> #include <linux/regmap.h> #include <sound/asoundef.h> @@ -1559,10 +1560,14 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) ret = snd_soc_register_component(&pdev->dev, &stm32_component, &sai->cpu_dai_drv, 1); - if (ret) + if (ret) { snd_dmaengine_pcm_unregister(&pdev->dev); + return ret; + } - return ret; + pm_runtime_enable(&pdev->dev); + + return 0; } static int stm32_sai_sub_remove(struct platform_device *pdev) @@ -1572,6 +1577,7 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) clk_unprepare(sai->pdata->pclk); snd_dmaengine_pcm_unregister(&pdev->dev); snd_soc_unregister_component(&pdev->dev); + pm_runtime_disable(&pdev->dev); return 0; } -- 2.17.1

2 1

ucm2: USB-Audio: Add CaptureMixerElem for Lenovo ThinkStation P620
by GitHub pull_request - opened 14 Sep '20

14 Sep '20

alsa-project/alsa-ucm-conf pull request #51 was opened from khfeng: The Switch and Volume mixer start to work after device firmware update. Add MixerElem to support them. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/51 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/51.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: ak4458: Add DSD support for ak4458 and ak4497
by Shengjiu Wang 14 Sep '20

14 Sep '20

ak4458 can't support DSD512 format, but ak4497 can, so add a new variable (dsd512) in ak4458_drvdata to distinguish these two platforms. Add a new kcontrol for ak4497 codec for ak4497 has a specific pin selection. In hw_params(), calculate bit clock according to different DSD format and configure DSD registers. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - add more commit description. sound/soc/codecs/ak4458.c | 82 +++++++++++++++++++++++++++++++++++---- sound/soc/codecs/ak4458.h | 5 ++- 2 files changed, 79 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 763e6839428f..2ef8b591eb79 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -31,11 +31,13 @@ static const char *ak4458_supply_names[AK4458_NUM_SUPPLIES] = { struct ak4458_drvdata { struct snd_soc_dai_driver *dai_drv; const struct snd_soc_component_driver *comp_drv; + bool dsd512; /* DSD512 is supported or not */ }; /* AK4458 Codec Private Data */ struct ak4458_priv { struct regulator_bulk_data supplies[AK4458_NUM_SUPPLIES]; + const struct ak4458_drvdata *drvdata; struct device *dev; struct regmap *regmap; struct gpio_desc *reset_gpiod; @@ -136,6 +138,10 @@ static const char * const ak4458_ats_select_texts[] = { /* DIF2 bit Audio Interface Format Setting(BICK fs) */ static const char * const ak4458_dif_select_texts[] = {"32fs,48fs", "64fs",}; +/* DSD input pin select */ +static const char * const ak4497_dsd_input_pin_select[] = { + "16_17_19pin", "3_4_5pin"}; + static const struct soc_enum ak4458_dac1_dem_enum = SOC_ENUM_SINGLE(AK4458_01_CONTROL2, 1, ARRAY_SIZE(ak4458_dem_select_texts), @@ -175,6 +181,10 @@ static const struct soc_enum ak4458_dif_enum = SOC_ENUM_SINGLE(AK4458_00_CONTROL1, 3, ARRAY_SIZE(ak4458_dif_select_texts), ak4458_dif_select_texts); +static const struct soc_enum ak4497_dsdp_enum = + SOC_ENUM_SINGLE(AK4458_09_DSD2, 2, + ARRAY_SIZE(ak4497_dsd_input_pin_select), + ak4497_dsd_input_pin_select); static int get_digfil(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -282,6 +292,7 @@ static const struct snd_kcontrol_new ak4497_snd_controls[] = { SOC_ENUM("AK4497 Sound Mode", ak4458_sm_enum), SOC_ENUM("AK4497 Attenuation transition Time Setting", ak4458_ats_enum), + SOC_ENUM("AK4497 DSD Data Input Pin", ak4497_dsdp_enum), }; /* ak4497 dapm widgets */ @@ -325,12 +336,54 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int pcm_width = max(params_physical_width(params), ak4458->slot_width); - int nfs1; - u8 format; + u8 format, dsdsel0, dsdsel1; + int nfs1, dsd_bclk; nfs1 = params_rate(params); ak4458->fs = nfs1; + /* calculate bit clock */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + dsd_bclk = nfs1 * params_physical_width(params); + switch (dsd_bclk) { + case 2822400: + dsdsel0 = 0; + dsdsel1 = 0; + break; + case 5644800: + dsdsel0 = 1; + dsdsel1 = 0; + break; + case 11289600: + dsdsel0 = 0; + dsdsel1 = 1; + break; + case 22579200: + if (ak4458->drvdata->dsd512) { + dsdsel0 = 1; + dsdsel1 = 1; + } else { + dev_err(dai->dev, "DSD512 not supported.\n"); + return -EINVAL; + } + break; + default: + dev_err(dai->dev, "Unsupported dsd bclk.\n"); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AK4458_06_DSD1, + AK4458_DSDSEL_MASK, dsdsel0); + snd_soc_component_update_bits(component, AK4458_09_DSD2, + AK4458_DSDSEL_MASK, dsdsel1); + break; + } + /* Master Clock Frequency Auto Setting Mode Enable */ snd_soc_component_update_bits(component, AK4458_00_CONTROL1, 0x80, 0x80); @@ -355,6 +408,9 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_DSP_B: format = AK4458_DIF_32BIT_MSB; break; + case SND_SOC_DAIFMT_PDM: + format = AK4458_DIF_32BIT_MSB; + break; default: return -EINVAL; } @@ -393,6 +449,7 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_PDM: ak4458->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; break; default: @@ -401,6 +458,12 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + /* DSD mode */ + snd_soc_component_update_bits(component, AK4458_02_CONTROL3, + AK4458_DP_MASK, + ak4458->fmt == SND_SOC_DAIFMT_PDM ? + AK4458_DP_MASK : 0); + ak4458_rstn_control(component, 0); ak4458_rstn_control(component, 1); @@ -472,7 +535,10 @@ static int ak4458_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, #define AK4458_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_DSD_U8 |\ + SNDRV_PCM_FMTBIT_DSD_U16_LE |\ + SNDRV_PCM_FMTBIT_DSD_U32_LE) static const unsigned int ak4458_rates[] = { 8000, 11025, 16000, 22050, @@ -668,11 +734,13 @@ static const struct regmap_config ak4458_regmap = { static const struct ak4458_drvdata ak4458_drvdata = { .dai_drv = &ak4458_dai, .comp_drv = &soc_codec_dev_ak4458, + .dsd512 = false, }; static const struct ak4458_drvdata ak4497_drvdata = { .dai_drv = &ak4497_dai, .comp_drv = &soc_codec_dev_ak4497, + .dsd512 = true, }; static const struct dev_pm_ops ak4458_pm = { @@ -684,7 +752,6 @@ static const struct dev_pm_ops ak4458_pm = { static int ak4458_i2c_probe(struct i2c_client *i2c) { struct ak4458_priv *ak4458; - const struct ak4458_drvdata *drvdata; int ret, i; ak4458 = devm_kzalloc(&i2c->dev, sizeof(*ak4458), GFP_KERNEL); @@ -698,7 +765,7 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) i2c_set_clientdata(i2c, ak4458); ak4458->dev = &i2c->dev; - drvdata = of_device_get_match_data(&i2c->dev); + ak4458->drvdata = of_device_get_match_data(&i2c->dev); ak4458->reset_gpiod = devm_gpiod_get_optional(ak4458->dev, "reset", GPIOD_OUT_LOW); @@ -720,8 +787,9 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) return ret; } - ret = devm_snd_soc_register_component(ak4458->dev, drvdata->comp_drv, - drvdata->dai_drv, 1); + ret = devm_snd_soc_register_component(ak4458->dev, + ak4458->drvdata->comp_drv, + ak4458->drvdata->dai_drv, 1); if (ret < 0) { dev_err(ak4458->dev, "Failed to register CODEC: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4458.h b/sound/soc/codecs/ak4458.h index f906215f7e4e..9548c5d78621 100644 --- a/sound/soc/codecs/ak4458.h +++ b/sound/soc/codecs/ak4458.h @@ -83,4 +83,7 @@ #define AK4458_ATS_SHIFT 6 #define AK4458_ATS_MASK GENMASK(7, 6) -#endif /* _AK4458_H */ +#define AK4458_DSDSEL_MASK (0x1 << 0) +#define AK4458_DP_MASK (0x1 << 7) + +#endif -- 2.27.0

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Re: [PATCH] ASoC: tas6424: disable regulators when probe error
by Mark Brown 14 Sep '20

14 Sep '20

On Sat, Sep 12, 2020 at 11:31:51AM +0900, Chanho Park wrote: > To fix up below errors when tas6424 chip is not available, this patch > disables regulators before returning error. > > [ 9.367234] tas6424 2-006a: unable to reset device: -6 > [ 9.499118] Call trace: > [ 9.501550] _regulator_put.part.21+0x158/0x160 > [ 9.506065] regulator_put+0x34/0x48 > [ 9.509624] regulator_bulk_free+0x30/0x50 > [ 9.513705] devm_regulator_bulk_release+0x18/0x20 > [ 9.518483] release_nodes+0x1b8/0x2a0 > [ 9.522215] devres_release_all+0x34/0x50 Please think hard before including complete backtraces in upstream reports, they are very large and contain almost no useful information relative to their size so often obscure the relevant content in your message. If part of the backtrace is usefully illustrative (it often is for search engines if nothing else) then it's usually better to pull out the relevant sections.

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control: snd_ctl_open(): which names are acceptable?
by Tanjeff Moos 13 Sep '20

13 Sep '20

Hello Alsa developers, I'm currently enhancing the libalsa API documentation. In the process, I try to understand how each API function works and what the underlying concepts are. The snd_ctl_open() call takes a name, which is a C string (const char*). For my sound card, the names "hw:CARD=PCH" and as well as "hw:2" (2 is the card index) worked. Which strings are acceptable as name? Are there maybe even more formats? Is there some documentation on this? Regards, Tanjeff

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[GIT PULL] ASoC fixes for v5.9-rc4
by Mark Brown 12 Sep '20

12 Sep '20

The following changes since commit d012a7190fc1fd72ed48911e77ca97ba4521bccd: Linux 5.9-rc2 (2020-08-23 14:08:43 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.9-rc4 for you to fetch changes up to 1a5ce48fd667128e369fdc7fb87e21539aed21b5: ASoC: tlv320adcx140: Wake up codec before accessing register (2020-09-08 17:26:03 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.9 Most of this is various driver specific fixes, none of which are terribly exciting in themselves, plus one core fix adding and using a new DAI lookup function to deal with a lockdep warning. ---------------------------------------------------------------- Brent Lu (1): ASoC: intel: atom: Add period size constraint Camel Guo (2): ASoC: tlv320adcx140: Fix accessing uninitialized adcx140->dev ASoC: tlv320adcx140: Wake up codec before accessing register Cezary Rojewski (2): ASoC: Intel: haswell: Fix power transition refactor ASoC: core: Do not cleanup uninitialized dais on soc_pcm_open failure Dinghao Liu (1): ASoC: qcom: common: Fix refcount imbalance on error Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add quirk for MPMAN Converter9 2-in-1 Jerome Brunet (1): ASoC: meson: axg-toddr: fix channel order on g12 platforms Kuninori Morimoto (3): ASoC: pcm3168a: ignore 0 Hz settings ASoC: ti: fixup ams_delta_mute() function name ASoC: soc-core: add snd_soc_find_dai_with_mutex() Mark Brown (2): Merge tag 'v5.9-rc2' into asoc-5.9 Merge series "ASoC: Fix return check for devm_regmap_init_sdw()" from Vinod Koul <vkoul(a)kernel.org>: Mateusz Gorski (1): ASoC: Intel: skl_hda_dsp_generic: Fix NULLptr dereference in autosuspend delay Rander Wang (1): ASoC: Intel: tgl_max98373: fix a runtime pm issue in multi-thread case Stephan Gerhold (1): ASoC: qcom: Set card->owner to avoid warnings Sylwester Nawrocki (2): ASoC: wm8994: Skip setting of the WM8994_MICBIAS register for WM1811 ASoC: wm8994: Ensure the device is resumed in wm89xx_mic_detect functions Vinod Koul (5): ASoC: max98373: Fix return check for devm_regmap_init_sdw() ASoC: rt1308-sdw: Fix return check for devm_regmap_init_sdw() ASoC: rt711: Fix return check for devm_regmap_init_sdw() ASoC: rt715: Fix return check for devm_regmap_init_sdw() ASoC: rt700: Fix return check for devm_regmap_init_sdw() include/sound/soc.h | 4 + sound/soc/codecs/max98373-sdw.c | 4 +- sound/soc/codecs/pcm3168a.c | 7 + sound/soc/codecs/rt1308-sdw.c | 4 +- sound/soc/codecs/rt700-sdw.c | 4 +- sound/soc/codecs/rt711-sdw.c | 4 +- sound/soc/codecs/rt715-sdw.c | 4 +- sound/soc/codecs/tlv320adcx140.c | 28 ++-- sound/soc/codecs/wm8994.c | 10 ++ sound/soc/codecs/wm_hubs.c | 3 + sound/soc/codecs/wm_hubs.h | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 11 ++ sound/soc/intel/boards/bytcr_rt5640.c | 10 ++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/boards/sof_maxim_common.c | 7 +- sound/soc/intel/haswell/sst-haswell-dsp.c | 185 ++++++++++++--------------- sound/soc/meson/axg-toddr.c | 24 +++- sound/soc/qcom/apq8016_sbc.c | 1 + sound/soc/qcom/apq8096.c | 1 + sound/soc/qcom/common.c | 6 +- sound/soc/qcom/sdm845.c | 1 + sound/soc/qcom/storm.c | 1 + sound/soc/soc-core.c | 13 ++ sound/soc/soc-dai.c | 4 +- sound/soc/soc-pcm.c | 2 +- sound/soc/ti/ams-delta.c | 4 +- 26 files changed, 206 insertions(+), 139 deletions(-)

2 1

[PATCH v10 0/3] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 11 Sep '20

11 Sep '20

Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. - This patch series needs HDMI DAI name SC7180_LPASS_DP defined in sc7180-lpass.h. It will be posted in the newer patchset of https://patchwork.kernel.org/patch/11745565/ Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Changes from v6 to v7 - Documentation: - Add headset-jack and hdmi-jack to specify the codec responsible for jack detection. - HDMI codec driver: - Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect. - Machine driver: - Removed aux device following Stephan's suggestion. - Use headset-jack and hdmi-jack to specify the codec responsible for jack detection. - Add support for HDMI(actually DP) playback. Changes from v7 to v8 - Documentation: - Remove headset-jack and hdmi-jack. - Machine driver: - Let machine driver decide whether there is a jack on the DAI. Changes from v8 to v9 - hdmi-codec driver: - Fixed the naming. - Machine driver: - Fixed unused fields. - Moved snd_soc_card_set_drvdata - Keep the naming of HDMI as dai name until v5 of lpass-hdmi patches. Changes from v9 to v10 - Documentation: - Let compatible string be more specific for board configuration to allow for future changes. - Machine driver: - Fixed unused include and macro. - Add temporary macro SC7180_LPASS_DP for future change in sc7180-lpass.h. - Let sound card be dynamically allocated. - Change compatible string accordingly. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (2): ASoC: hdmi-codec: Use set_jack ops to set jack ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 130 +++++++++ include/sound/hdmi-codec.h | 3 - sound/soc/codecs/hdmi-codec.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 5 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 +- sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 266 ++++++++++++++++++ sound/soc/rockchip/rockchip_max98090.c | 3 +- 10 files changed, 421 insertions(+), 22 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.618.gf4bc123cb7-goog

1 3

[PATCH v9 0/3] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 11 Sep '20

11 Sep '20

Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. - This patch series needs HDMI DAI name defined in sc7180-lpass.h. https://patchwork.kernel.org/patch/11745565/ Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Changes from v6 to v7 - Documentation: - Add headset-jack and hdmi-jack to specify the codec responsible for jack detection. - HDMI codec driver: - Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect. - Machine driver: - Removed aux device following Stephan's suggestion. - Use headset-jack and hdmi-jack to specify the codec responsible for jack detection. - Add support for HDMI(actually DP) playback. Changes from v7 to v8 - Documentation: - Remove headset-jack and hdmi-jack. - Machine driver: - Let machine driver decide whether there is a jack on the DAI. Changes from v8 to v9 - hdmi-codec driver: - Fixed the naming. - Machine driver: - Fixed unused fields. - Moved snd_soc_card_set_drvdata - Keep the naming of HDMI as dai name until v5 of lpass-hdmi patches. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (2): ASoC: hdmi-codec: Use set_jack ops to set jack ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 130 +++++++++ include/sound/hdmi-codec.h | 3 - sound/soc/codecs/hdmi-codec.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 5 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 +- sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 267 ++++++++++++++++++ sound/soc/rockchip/rockchip_max98090.c | 3 +- 10 files changed, 422 insertions(+), 22 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.618.gf4bc123cb7-goog

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[PATCH v2] ALSA: firewire: Replace tasklet with work
by Takashi Iwai 11 Sep '20

11 Sep '20

The tasklet is an old API that should be deprecated, usually can be converted to another decent API. In FireWire driver, a tasklet is still used for offloading the AMDTP PCM stream handling. It can be achieved gracefully with a work queued, too. This patch replaces the tasklet usage in firewire-lib driver with a simple work. The conversion is fairly straightforward but for the in_interrupt() checks that are replaced with the check using the current_work(). Note that in_interrupt() in amdtp_packet tracepoint is still kept as is. This is the place that is probed by both softirq of 1394 OHCI and a user task of a PCM application, and the work handling is already filtered in amdtp_domain_stream_pcm_pointer(). Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Drop in_interrupt() conversion in tracepoint sound/firewire/amdtp-stream.c | 25 +++++++++++++------------ sound/firewire/amdtp-stream.h | 2 +- 2 files changed, 14 insertions(+), 13 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index ee1c428b1fd3..4e2f2bb7879f 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -64,7 +64,7 @@ #define IT_PKT_HEADER_SIZE_CIP 8 // For 2 CIP header. #define IT_PKT_HEADER_SIZE_NO_CIP 0 // Nothing. -static void pcm_period_tasklet(struct tasklet_struct *t); +static void pcm_period_work(struct work_struct *work); /** * amdtp_stream_init - initialize an AMDTP stream structure @@ -94,7 +94,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->flags = flags; s->context = ERR_PTR(-1); mutex_init(&s->mutex); - tasklet_setup(&s->period_tasklet, pcm_period_tasklet); + INIT_WORK(&s->period_work, pcm_period_work); s->packet_index = 0; init_waitqueue_head(&s->callback_wait); @@ -203,7 +203,7 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s, // Linux driver for 1394 OHCI controller voluntarily flushes isoc // context when total size of accumulated context header reaches - // PAGE_SIZE. This kicks tasklet for the isoc context and brings + // PAGE_SIZE. This kicks work for the isoc context and brings // callback in the middle of scheduled interrupts. // Although AMDTP streams in the same domain use the same events per // IRQ, use the largest size of context header between IT/IR contexts. @@ -333,7 +333,7 @@ EXPORT_SYMBOL(amdtp_stream_get_max_payload); */ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) { - tasklet_kill(&s->period_tasklet); + cancel_work_sync(&s->period_work); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; } @@ -437,13 +437,14 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - tasklet_hi_schedule(&s->period_tasklet); + queue_work(system_highpri_wq, &s->period_work); } } -static void pcm_period_tasklet(struct tasklet_struct *t) +static void pcm_period_work(struct work_struct *work) { - struct amdtp_stream *s = from_tasklet(s, t, period_tasklet); + struct amdtp_stream *s = container_of(work, struct amdtp_stream, + period_work); struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); if (pcm) @@ -794,7 +795,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, struct pkt_desc *descs, static inline void cancel_stream(struct amdtp_stream *s) { s->packet_index = -1; - if (in_interrupt()) + if (current_work() == &s->period_work) amdtp_stream_pcm_abort(s); WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN); } @@ -1184,7 +1185,7 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, if (irq_target && amdtp_stream_running(irq_target)) { // This function is called in software IRQ context of - // period_tasklet or process context. + // period_work or process context. // // When the software IRQ context was scheduled by software IRQ // context of IT contexts, queued packets were already handled. @@ -1195,9 +1196,9 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d, // immediately to keep better granularity of PCM pointer. // // Later, the process context will sometimes schedules software - // IRQ context of the period_tasklet. Then, no need to flush the + // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above - if (!in_interrupt()) { + if (current_work() != &s->period_work) { // Queued packet should be processed without any kernel // preemption to keep latency against bus cycle. preempt_disable(); @@ -1263,7 +1264,7 @@ static void amdtp_stream_stop(struct amdtp_stream *s) return; } - tasklet_kill(&s->period_tasklet); + cancel_work_sync(&s->period_work); fw_iso_context_stop(s->context); fw_iso_context_destroy(s->context); s->context = ERR_PTR(-1); diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 703b710aaf7f..2ceb57d1d58e 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -163,7 +163,7 @@ struct amdtp_stream { /* For a PCM substream processing. */ struct snd_pcm_substream *pcm; - struct tasklet_struct period_tasklet; + struct work_struct period_work; snd_pcm_uframes_t pcm_buffer_pointer; unsigned int pcm_period_pointer; -- 2.16.4

2 2

[PATCH 1/4] ASoC: odroid: Add missing properties
by Krzysztof Kozlowski 11 Sep '20

11 Sep '20

Usage of "unevaluatedProperties: false" is not correct as it suppresses warnings about all undocumented properties. Instead, add all missing properties. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Fixup for: https://lore.kernel.org/lkml/30b8ea43-c2d0-5ddf-dc16-a8fe80d47c38@samsung.c… --- .../devicetree/bindings/sound/samsung,odroid.yaml | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml index de1be3d6d1e9..e8122bc87362 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -28,6 +28,11 @@ properties: $ref: /schemas/types.yaml#/definitions/string description: The user-visible name of this sound complex. + assigned-clock-parents: true + assigned-clock-rates: true + assigned-clocks: true + clocks: true + cpu: type: object properties: @@ -69,7 +74,7 @@ required: - cpu - codec -unevaluatedProperties: false +additionalProperties: false examples: - | -- 2.17.1

2 5

[PATCH] ASoC: core: Optionaly add the component driver name to the component name
by Dafna Hirschfeld 11 Sep '20

11 Sep '20

The ASoC platform driver for Mediatek MT8173 registers several components with identical name. This cause an error: "Directory '11220000.audio-controller' with parent 'mtk-rt5650' already present!" when adding debugfs directory for those components. To fix this, the function 'fmt_single_name' can receive the name of the component driver and concatenate it to the name of the debugfs directory for the component. Signed-off-by: Dafna Hirschfeld <dafna.hirschfeld(a)collabora.com> --- sound/soc/soc-core.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 74df22486e30..f470e713b030 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2227,7 +2227,7 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_card); * Simplify DAI link configuration by removing ".-1" from device names * and sanitizing names. */ -static char *fmt_single_name(struct device *dev, int *id) +static char *fmt_single_name(struct device *dev, int *id, const char *component_driver_name) { const char *devname = dev_name(dev); char *found, *name; @@ -2263,6 +2263,12 @@ static char *fmt_single_name(struct device *dev, int *id) *id = 0; } + if (component_driver_name) { + char *tmp = name; + + name = devm_kasprintf(dev, GFP_KERNEL, "%s-%s", name, component_driver_name); + devm_kfree(dev, tmp); + } return name; } @@ -2327,7 +2333,7 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, */ if (legacy_dai_naming && (dai_drv->id == 0 || dai_drv->name == NULL)) { - dai->name = fmt_single_name(dev, &dai->id); + dai->name = fmt_single_name(dev, &dai->id, NULL); } else { dai->name = fmt_multiple_name(dev, dai_drv); if (dai_drv->id) @@ -2460,7 +2466,7 @@ int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->card_list); mutex_init(&component->io_mutex); - component->name = fmt_single_name(dev, &component->id); + component->name = fmt_single_name(dev, &component->id, driver->name); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.17.1

2 1

[PATCH] ASoC: ak4458: Add DSD support for ak4458 and ak4497
by Shengjiu Wang 11 Sep '20

11 Sep '20

ak4458 can't support DSD512 format, but ak4497 can, so add a new variable in ak4458_drvdata to distinguish these two platform. In hw_params(), calculate bit clock according to different DSD format and configure DSD register. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/ak4458.c | 82 +++++++++++++++++++++++++++++++++++---- sound/soc/codecs/ak4458.h | 5 ++- 2 files changed, 79 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 763e6839428f..9054964d5cfa 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -31,11 +31,13 @@ static const char *ak4458_supply_names[AK4458_NUM_SUPPLIES] = { struct ak4458_drvdata { struct snd_soc_dai_driver *dai_drv; const struct snd_soc_component_driver *comp_drv; + bool dsd512; /* DSD512 is supported or not */ }; /* AK4458 Codec Private Data */ struct ak4458_priv { struct regulator_bulk_data supplies[AK4458_NUM_SUPPLIES]; + const struct ak4458_drvdata *drvdata; struct device *dev; struct regmap *regmap; struct gpio_desc *reset_gpiod; @@ -136,6 +138,10 @@ static const char * const ak4458_ats_select_texts[] = { /* DIF2 bit Audio Interface Format Setting(BICK fs) */ static const char * const ak4458_dif_select_texts[] = {"32fs,48fs", "64fs",}; +/* DSD input pin select */ +static const char * const ak4497_dsd_input_path_select[] = { + "16_17_19pin", "3_4_5pin"}; + static const struct soc_enum ak4458_dac1_dem_enum = SOC_ENUM_SINGLE(AK4458_01_CONTROL2, 1, ARRAY_SIZE(ak4458_dem_select_texts), @@ -175,6 +181,10 @@ static const struct soc_enum ak4458_dif_enum = SOC_ENUM_SINGLE(AK4458_00_CONTROL1, 3, ARRAY_SIZE(ak4458_dif_select_texts), ak4458_dif_select_texts); +static const struct soc_enum ak4497_dsdp_enum = + SOC_ENUM_SINGLE(AK4458_09_DSD2, 2, + ARRAY_SIZE(ak4497_dsd_input_path_select), + ak4497_dsd_input_path_select); static int get_digfil(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -282,6 +292,7 @@ static const struct snd_kcontrol_new ak4497_snd_controls[] = { SOC_ENUM("AK4497 Sound Mode", ak4458_sm_enum), SOC_ENUM("AK4497 Attenuation transition Time Setting", ak4458_ats_enum), + SOC_ENUM("AK4497 DSD Data Input Pin", ak4497_dsdp_enum), }; /* ak4497 dapm widgets */ @@ -325,12 +336,54 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int pcm_width = max(params_physical_width(params), ak4458->slot_width); - int nfs1; - u8 format; + u8 format, dsdsel0, dsdsel1; + int nfs1, dsd_bclk; nfs1 = params_rate(params); ak4458->fs = nfs1; + /* calculate bit clock */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_DSD_U8: + case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U16_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: + case SNDRV_PCM_FORMAT_DSD_U32_BE: + dsd_bclk = nfs1 * params_physical_width(params); + switch (dsd_bclk) { + case 2822400: + dsdsel0 = 0; + dsdsel1 = 0; + break; + case 5644800: + dsdsel0 = 1; + dsdsel1 = 0; + break; + case 11289600: + dsdsel0 = 0; + dsdsel1 = 1; + break; + case 22579200: + if (ak4458->drvdata->dsd512) { + dsdsel0 = 1; + dsdsel1 = 1; + } else { + dev_err(dai->dev, "DSD512 not supported.\n"); + return -EINVAL; + } + break; + default: + dev_err(dai->dev, "Unsupported dsd bclk.\n"); + return -EINVAL; + } + + snd_soc_component_update_bits(component, AK4458_06_DSD1, + AK4458_DSDSEL_MASK, dsdsel0); + snd_soc_component_update_bits(component, AK4458_09_DSD2, + AK4458_DSDSEL_MASK, dsdsel1); + break; + } + /* Master Clock Frequency Auto Setting Mode Enable */ snd_soc_component_update_bits(component, AK4458_00_CONTROL1, 0x80, 0x80); @@ -355,6 +408,9 @@ static int ak4458_hw_params(struct snd_pcm_substream *substream, case SND_SOC_DAIFMT_DSP_B: format = AK4458_DIF_32BIT_MSB; break; + case SND_SOC_DAIFMT_PDM: + format = AK4458_DIF_32BIT_MSB; + break; default: return -EINVAL; } @@ -393,6 +449,7 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_LEFT_J: case SND_SOC_DAIFMT_RIGHT_J: case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_PDM: ak4458->fmt = fmt & SND_SOC_DAIFMT_FORMAT_MASK; break; default: @@ -401,6 +458,12 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } + /* DSD mode */ + snd_soc_component_update_bits(component, AK4458_02_CONTROL3, + AK4458_DP_MASK, + ak4458->fmt == SND_SOC_DAIFMT_PDM ? + AK4458_DP_MASK : 0); + ak4458_rstn_control(component, 0); ak4458_rstn_control(component, 1); @@ -472,7 +535,10 @@ static int ak4458_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, #define AK4458_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S32_LE |\ + SNDRV_PCM_FMTBIT_DSD_U8 |\ + SNDRV_PCM_FMTBIT_DSD_U16_LE |\ + SNDRV_PCM_FMTBIT_DSD_U32_LE) static const unsigned int ak4458_rates[] = { 8000, 11025, 16000, 22050, @@ -668,11 +734,13 @@ static const struct regmap_config ak4458_regmap = { static const struct ak4458_drvdata ak4458_drvdata = { .dai_drv = &ak4458_dai, .comp_drv = &soc_codec_dev_ak4458, + .dsd512 = false, }; static const struct ak4458_drvdata ak4497_drvdata = { .dai_drv = &ak4497_dai, .comp_drv = &soc_codec_dev_ak4497, + .dsd512 = true, }; static const struct dev_pm_ops ak4458_pm = { @@ -684,7 +752,6 @@ static const struct dev_pm_ops ak4458_pm = { static int ak4458_i2c_probe(struct i2c_client *i2c) { struct ak4458_priv *ak4458; - const struct ak4458_drvdata *drvdata; int ret, i; ak4458 = devm_kzalloc(&i2c->dev, sizeof(*ak4458), GFP_KERNEL); @@ -698,7 +765,7 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) i2c_set_clientdata(i2c, ak4458); ak4458->dev = &i2c->dev; - drvdata = of_device_get_match_data(&i2c->dev); + ak4458->drvdata = of_device_get_match_data(&i2c->dev); ak4458->reset_gpiod = devm_gpiod_get_optional(ak4458->dev, "reset", GPIOD_OUT_LOW); @@ -720,8 +787,9 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) return ret; } - ret = devm_snd_soc_register_component(ak4458->dev, drvdata->comp_drv, - drvdata->dai_drv, 1); + ret = devm_snd_soc_register_component(ak4458->dev, + ak4458->drvdata->comp_drv, + ak4458->drvdata->dai_drv, 1); if (ret < 0) { dev_err(ak4458->dev, "Failed to register CODEC: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ak4458.h b/sound/soc/codecs/ak4458.h index f906215f7e4e..9548c5d78621 100644 --- a/sound/soc/codecs/ak4458.h +++ b/sound/soc/codecs/ak4458.h @@ -83,4 +83,7 @@ #define AK4458_ATS_SHIFT 6 #define AK4458_ATS_MASK GENMASK(7, 6) -#endif /* _AK4458_H */ +#define AK4458_DSDSEL_MASK (0x1 << 0) +#define AK4458_DP_MASK (0x1 << 7) + +#endif -- 2.27.0

3 2

[PATCH v8 0/3] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 11 Sep '20

11 Sep '20

Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. - This patch series needs HDMI DAI name defined in sc7180-lpass.h. https://patchwork.kernel.org/patch/11745565/ Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Changes from v6 to v7 - Documentation: - Add headset-jack and hdmi-jack to specify the codec responsible for jack detection. - HDMI codec driver: - Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect. - Machine driver: - Removed aux device following Stephan's suggestion. - Use headset-jack and hdmi-jack to specify the codec responsible for jack detection. - Add support for HDMI(actually DP) playback. Changes from v7 to v8 - Documentation: - Remove headset-jack and hdmi-jack. - Machine driver: - Let machine driver decide whether there is a jack on the DAI. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (2): ASoC: hdmi-codec: Use set_jack ops to set jack ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 130 +++++++++ include/sound/hdmi-codec.h | 3 - sound/soc/codecs/hdmi-codec.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 5 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 +- sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 270 ++++++++++++++++++ sound/soc/rockchip/rockchip_max98090.c | 3 +- 10 files changed, 425 insertions(+), 22 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.526.ge36021eeef-goog

2 5

[PATCH] soundwire: bus: add enumerated slave to device list
by Srinivas Kandagatla 11 Sep '20

11 Sep '20

Currently slave devices are only added either from device tree or acpi entries. However lets say, there is wrong or no entry of a slave device in DT that is enumerated, then there is no way for user to know all the enumerated devices on the bus. To fix this add slave device by default if there is no matching dt or acpi entry, so that we can see this in sysfs entry. In my case I had a wrong address entry in DT, However I had no way to know what devices are actually enumerated on the bus! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/bus.c | 1 + drivers/soundwire/bus.h | 2 ++ drivers/soundwire/bus_type.c | 6 ++++++ drivers/soundwire/slave.c | 4 ++-- 4 files changed, 11 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index e6e0fb9a81b4..55d9c22c4ec5 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -699,6 +699,7 @@ static int sdw_program_device_num(struct sdw_bus *bus) if (!found) { /* TODO: Park this device in Group 13 */ + sdw_slave_add(bus, &id, NULL); dev_err(bus->dev, "Slave Entry not found\n"); } diff --git a/drivers/soundwire/bus.h b/drivers/soundwire/bus.h index 82484f741168..1517d6789dff 100644 --- a/drivers/soundwire/bus.h +++ b/drivers/soundwire/bus.h @@ -19,6 +19,8 @@ static inline int sdw_acpi_find_slaves(struct sdw_bus *bus) int sdw_of_find_slaves(struct sdw_bus *bus); void sdw_extract_slave_id(struct sdw_bus *bus, u64 addr, struct sdw_slave_id *id); +int sdw_slave_add(struct sdw_bus *bus, struct sdw_slave_id *id, + struct fwnode_handle *fwnode); int sdw_master_device_add(struct sdw_bus *bus, struct device *parent, struct fwnode_handle *fwnode); int sdw_master_device_del(struct sdw_bus *bus); diff --git a/drivers/soundwire/bus_type.c b/drivers/soundwire/bus_type.c index 6fba55898cf0..ac036223046f 100644 --- a/drivers/soundwire/bus_type.c +++ b/drivers/soundwire/bus_type.c @@ -84,6 +84,12 @@ static int sdw_drv_probe(struct device *dev) const struct sdw_device_id *id; int ret; + /* + * fw description is mandatory to bind + */ + if (!dev->fwnode || !dev->of_node) + return -ENODEV; + id = sdw_get_device_id(slave, drv); if (!id) return -ENODEV; diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index 0839445ee07b..24a16ebf9ae2 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -20,8 +20,8 @@ struct device_type sdw_slave_type = { .uevent = sdw_slave_uevent, }; -static int sdw_slave_add(struct sdw_bus *bus, - struct sdw_slave_id *id, struct fwnode_handle *fwnode) +int sdw_slave_add(struct sdw_bus *bus, + struct sdw_slave_id *id, struct fwnode_handle *fwnode) { struct sdw_slave *slave; int ret; -- 2.21.0

3 8

[PATCH v2 0/2] ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation
by Kuninori Morimoto 10 Sep '20

10 Sep '20

Hi Rob, Mark These are v2 of switch to yaml base Documentation for audio-graph-card and renesas,rsnd which uses audio-graph-card. v1 -> v2 - upgrade dt-schema, and fixup errors for [2/2] patch Kuninori Morimoto (2): ASoC: dt-bindings: audio-graph-card: switch to yaml base Documentation ASoC: dt-bindings: renesas,rsnd: switch to yaml base Documentation .../bindings/sound/audio-graph-card.txt | 337 ------------ .../bindings/sound/audio-graph-card.yaml | 135 +++++ .../bindings/sound/renesas,rsnd.txt | 520 ------------------ .../bindings/sound/renesas,rsnd.yaml | 454 +++++++++++++++ .../devicetree/bindings/vendor-prefixes.yaml | 2 +- 5 files changed, 590 insertions(+), 858 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/audio-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml -- 2.25.1

2 6

[PATCH V6] ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88125_* machine
by Radoslaw Biernacki 10 Sep '20

10 Sep '20

Since 256xFS clocks cannot be generated by SKL, the NAU8825 is configured to re-generate its system clock from the BCLK using the FLL. The link is configured to use a 48kHz frame rate, and 24 bits in 25-bit slot. The SSP configuration is extracted from NHLT settings and not dynamically changed. Listening tests and measurements do not show any distortion or issues Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Radoslaw Biernacki <rad(a)semihalf.com> --- Notes: v1 -> v2: - adding same changes to skl_nau88l25_max98357a.c v2 -> v3: - removing msleep() in SNDRV_PCM_TRIGGER_RESUME as it unnecessarily increase playback/capture latency while actually FLL does not require it. - simplifing commit message v3 -> v4: - simplifing the PM resume callback code for setting the FLL - adding comment for the stream START/RESUME sequence which prevent audio pops - fixing mising var initialization in platform_clock_control() v4 -> v5: - removed stray change ~ SSP0 ops v5 -> v6: - fixing typos and comment style .../soc/intel/boards/skl_nau88l25_max98357a.c | 62 +++++++++++------- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 63 ++++++++++++------- 2 files changed, 83 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index d7b8154c43a4..12595138f44b 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -47,12 +48,12 @@ enum { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct snd_soc_dai *codec_dai; - int ret; + int ret = 0; codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); if (!codec_dai) { @@ -60,14 +61,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -292,24 +286,50 @@ static const struct snd_soc_ops skylake_nau8825_fe_ops = { .startup = skl_fe_startup, }; -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Since 256xFS clocks cannot be generated by SKL, the NAU8825 + * is configured to re-generate its system clock from the BCLK + * using the FLL. + * We must switch system clock (FLL to use BCLK) here as it is + * not given earlier by FW (like in hw_param). We let nau8825 to + * use internal VCO clock till now which reduces the audible + * pop's. + */ + case SNDRV_PCM_TRIGGER_RESUME: + /* Once device resumes, the system will only enable power + * sequence for playback without doing hardware parameter, audio + * format, and PLL configure. In the mean time, the jack + * detection sequence has changed PLL parameters and switched to + * internal clock. Thus, the playback signal distorted without + * correct PLL parameters. Therefore we need to configure PLL + * again + */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 4b317bcf6ea0..38faaf20306e 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -57,12 +58,12 @@ static const struct snd_kcontrol_new skylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct snd_soc_dai *codec_dai; - int ret; + int ret = 0; codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); if (!codec_dai) { @@ -70,14 +71,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -85,6 +79,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } } + return ret; } @@ -344,24 +339,50 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Since 256xFS clocks cannot be generated by SKL, the NAU8825 + * is configured to re-generate its system clock from the BCLK + * using the FLL. + * We must switch system clock (FLL to use BCLK) here as it is + * not given earlier by FW (like in hw_param). We let nau8825 to + * use internal VCO clock till now which reduces the audible + * pop's. + */ + case SNDRV_PCM_TRIGGER_RESUME: + /* Once device resumes, the system will only enable power + * sequence for playback without doing hardware parameter, audio + * format, and PLL configure. In the mean time, the jack + * detection sequence has changed PLL parameters and switched to + * internal clock. Thus, the playback signal distorted without + * correct PLL parameters. Therefore we need to configure PLL + * again + */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static const unsigned int channels_dmic[] = { -- 2.17.1

2 1

[PATCH V4] ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88125_* machine
by Radoslaw Biernacki 10 Sep '20

10 Sep '20

Since 256xFS clocks cannot be generated by SKL, the NAU8825 is configured to re-generate its system clock from the BCLK using the FLL. The link is configured to use a 48kHz frame rate, and 24 bits in 25-bit slot. The SSP configuration is extracted from NHLT settings and not dynamically changed. Listening tests and measurements do not show any distortion or issues Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Radoslaw Biernacki <rad(a)semihalf.com> --- Notes: v1 -> v2: - adding same changes to skl_nau88l25_max98357a.c v2 -> v3: - removing msleep() in SNDRV_PCM_TRIGGER_RESUME as it unnecessarily increase playback/capture latency while actually FLL does not require it. - simplifing commit message v3 -> v4: - simplifing the PM resume callback code for setting the FLL - adding comment for the stream START/RESUME sequence which prevent audio pops - fixing mising var initialization in platform_clock_control() .../soc/intel/boards/skl_nau88l25_max98357a.c | 63 ++++++++++++------ sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 65 +++++++++++++------ 2 files changed, 86 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index d7b8154c43a4..2f0abbd2dd8d 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -47,12 +48,12 @@ enum { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct snd_soc_dai *codec_dai; - int ret; + int ret = 0; codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); if (!codec_dai) { @@ -60,14 +61,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -292,24 +286,51 @@ static const struct snd_soc_ops skylake_nau8825_fe_ops = { .startup = skl_fe_startup, }; -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Since 256xFS clocks cannot be generated by SKL, the NAU8825 + * is configured to re-generate its system clock from the BCLK + * using the FLL. + * We must switch system clock (FLL to use BCLK) here as it is + * not given eariler by FW (like in hw_param). We let nau8825 to + * use internal VCO clock till now which reduces the audiable + * pop's. */ + + /* fall through */ + + case SNDRV_PCM_TRIGGER_RESUME: + /* Once device resumes, the system will only enable power + * sequence for playback without doing hardware parameter, audio + * format, and PLL configure. In the mean time, the jack + * detecion sequence has changed PLL parameters and switched to + * internal clock. Thus, the playback signal distorted without + * correct PLL parameters. Therefore we need to configure PLL + * again */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 4b317bcf6ea0..d076f19f9b78 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -57,12 +58,12 @@ static const struct snd_kcontrol_new skylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct snd_soc_dai *codec_dai; - int ret; + int ret = 0; codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); if (!codec_dai) { @@ -70,14 +71,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -85,6 +79,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } } + return ret; } @@ -344,24 +339,51 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Since 256xFS clocks cannot be generated by SKL, the NAU8825 + * is configured to re-generate its system clock from the BCLK + * using the FLL. + * We must switch system clock (FLL to use BCLK) here as it is + * not given eariler by FW (like in hw_param). We let nau8825 to + * use internal VCO clock till now which reduces the audiable + * pop's. */ + + /* fall through */ + + case SNDRV_PCM_TRIGGER_RESUME: + /* Once device resumes, the system will only enable power + * sequence for playback without doing hardware parameter, audio + * format, and PLL configure. In the mean time, the jack + * detecion sequence has changed PLL parameters and switched to + * internal clock. Thus, the playback signal distorted without + * correct PLL parameters. Therefore we need to configure PLL + * again */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static const unsigned int channels_dmic[] = { @@ -582,6 +604,7 @@ static struct snd_soc_dai_link skylake_dais[] = { .init = skylake_ssm4567_codec_init, .ignore_pmdown_time = 1, .be_hw_params_fixup = skylake_ssp_fixup, + .ops = &skylake_nau8825_ops, .dpcm_playback = 1, .dpcm_capture = 1, SND_SOC_DAILINK_REG(ssp0_pin, ssp0_codec, platform), -- 2.17.1

3 3

[PATCH V5] ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88125_* machine
by Radoslaw Biernacki 10 Sep '20

10 Sep '20

Since 256xFS clocks cannot be generated by SKL, the NAU8825 is configured to re-generate its system clock from the BCLK using the FLL. The link is configured to use a 48kHz frame rate, and 24 bits in 25-bit slot. The SSP configuration is extracted from NHLT settings and not dynamically changed. Listening tests and measurements do not show any distortion or issues Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Radoslaw Biernacki <rad(a)semihalf.com> --- Notes: v1 -> v2: - adding same changes to skl_nau88l25_max98357a.c v2 -> v3: - removing msleep() in SNDRV_PCM_TRIGGER_RESUME as it unnecessarily increase playback/capture latency while actually FLL does not require it. - simplifing commit message v3 -> v4: - simplifing the PM resume callback code for setting the FLL - adding comment for the stream START/RESUME sequence which prevent audio pops - fixing mising var initialization in platform_clock_control() V4 -> V5: - removed stray change ~ SSP0 ops .../soc/intel/boards/skl_nau88l25_max98357a.c | 63 ++++++++++++------ sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 64 +++++++++++++------ 2 files changed, 85 insertions(+), 42 deletions(-) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index d7b8154c43a4..2f0abbd2dd8d 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -47,12 +48,12 @@ enum { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct snd_soc_dai *codec_dai; - int ret; + int ret = 0; codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); if (!codec_dai) { @@ -60,14 +61,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -292,24 +286,51 @@ static const struct snd_soc_ops skylake_nau8825_fe_ops = { .startup = skl_fe_startup, }; -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Since 256xFS clocks cannot be generated by SKL, the NAU8825 + * is configured to re-generate its system clock from the BCLK + * using the FLL. + * We must switch system clock (FLL to use BCLK) here as it is + * not given eariler by FW (like in hw_param). We let nau8825 to + * use internal VCO clock till now which reduces the audiable + * pop's. */ + + /* fall through */ + + case SNDRV_PCM_TRIGGER_RESUME: + /* Once device resumes, the system will only enable power + * sequence for playback without doing hardware parameter, audio + * format, and PLL configure. In the mean time, the jack + * detecion sequence has changed PLL parameters and switched to + * internal clock. Thus, the playback signal distorted without + * correct PLL parameters. Therefore we need to configure PLL + * again */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 4b317bcf6ea0..6791be63c866 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -57,12 +58,12 @@ static const struct snd_kcontrol_new skylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; struct snd_soc_dai *codec_dai; - int ret; + int ret = 0; codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); if (!codec_dai) { @@ -70,14 +71,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -85,6 +79,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } } + return ret; } @@ -344,24 +339,51 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Since 256xFS clocks cannot be generated by SKL, the NAU8825 + * is configured to re-generate its system clock from the BCLK + * using the FLL. + * We must switch system clock (FLL to use BCLK) here as it is + * not given eariler by FW (like in hw_param). We let nau8825 to + * use internal VCO clock till now which reduces the audiable + * pop's. */ + + /* fall through */ + + case SNDRV_PCM_TRIGGER_RESUME: + /* Once device resumes, the system will only enable power + * sequence for playback without doing hardware parameter, audio + * format, and PLL configure. In the mean time, the jack + * detecion sequence has changed PLL parameters and switched to + * internal clock. Thus, the playback signal distorted without + * correct PLL parameters. Therefore we need to configure PLL + * again */ + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static const unsigned int channels_dmic[] = { -- 2.17.1

1 0

[PATCH V3] ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88125_* machine
by Radoslaw Biernacki 10 Sep '20

10 Sep '20

Since 256xFS clocks cannot be generated by SKL, the NAU8825 is configured to re-generate its system clock from the BCLK using the FLL. The link is configured to use a 48kHz frame rate, and 24 bits in 25-bit slot. The SSP configuration is extracted from NHLT settings and not dynamically changed. Listening tests and measurements do not show any distortion or issues Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Radoslaw Biernacki <rad(a)semihalf.com> --- Notes: v1 -> v2: - adding same changes to skl_nau88l25_max98357a.c v2 -> v3: - removing msleep() in SNDRV_PCM_TRIGGER_RESUME as it unnecessarily increase playback/capture latency while actually FLL does not require it. - simplifing commit message .../soc/intel/boards/skl_nau88l25_max98357a.c | 71 +++++++++++++----- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 72 ++++++++++++++----- 2 files changed, 105 insertions(+), 38 deletions(-) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index d7b8154c43a4..75976aad0366 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -47,7 +48,7 @@ enum { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; @@ -60,14 +61,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -292,24 +286,39 @@ static const struct snd_soc_ops skylake_nau8825_fe_ops = { .startup = skl_fe_startup, }; -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + int ret = 0; - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + case SNDRV_PCM_TRIGGER_RESUME: + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, @@ -630,10 +639,34 @@ static int skylake_card_late_probe(struct snd_soc_card *card) return hdac_hdmi_jack_port_init(component, &card->dapm); } +static int __maybe_unused skylake_nau8825_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EIO; + } + + dev_dbg(codec_dai->dev, "playback_active:%d playback_widget->active:%d codec_dai->rate:%d\n", + codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], + codec_dai->playback_widget->active, + codec_dai->rate); + + if (codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] && + codec_dai->playback_widget->active) + snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825max", .owner = THIS_MODULE, + .resume_post = skylake_nau8825_resume_post, .dai_link = skylake_dais, .num_links = ARRAY_SIZE(skylake_dais), .controls = skylake_controls, diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 4b317bcf6ea0..09f090ba7fec 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -57,7 +58,7 @@ static const struct snd_kcontrol_new skylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; @@ -70,14 +71,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -85,6 +79,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } } + return ret; } @@ -344,24 +339,39 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + int ret = 0; - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + case SNDRV_PCM_TRIGGER_RESUME: + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static const unsigned int channels_dmic[] = { @@ -671,10 +681,34 @@ static int skylake_card_late_probe(struct snd_soc_card *card) return hdac_hdmi_jack_port_init(component, &card->dapm); } +static int __maybe_unused skylake_nau8825_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EIO; + } + + dev_dbg(codec_dai->dev, "playback_active:%d playback_widget->active:%d codec_dai->rate:%d\n", + codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], + codec_dai->playback_widget->active, + codec_dai->rate); + + if (codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] && + codec_dai->playback_widget->active) + snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825adi", .owner = THIS_MODULE, + .resume_post = skylake_nau8825_resume_post, .dai_link = skylake_dais, .num_links = ARRAY_SIZE(skylake_dais), .controls = skylake_controls, -- 2.17.1

3 2

[PATCH] soundwire: add slave status as a sysfs file
by Vinod Koul 10 Sep '20

10 Sep '20

SoundWire Slave status is not reported in the sysfs, so add this file with string values for status to userspace. Users can look into this file to see the Slave status at that point of time. Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- On RB3 it shows as: root@linaro-alip:/sys/bus/soundwire/devices# cat sdw:0:217:2010:0:1/status Attached root@linaro-alip:/sys/bus/soundwire/devices# cat sdw:0:217:2010:0:2/status Attached .../ABI/testing/sysfs-bus-soundwire-slave | 7 +++++++ drivers/soundwire/sysfs_slave.c | 17 +++++++++++++++++ include/linux/soundwire/sdw.h | 2 ++ 3 files changed, 26 insertions(+) diff --git a/Documentation/ABI/testing/sysfs-bus-soundwire-slave b/Documentation/ABI/testing/sysfs-bus-soundwire-slave index db4c9511d1aa..8a64f8e9079a 100644 --- a/Documentation/ABI/testing/sysfs-bus-soundwire-slave +++ b/Documentation/ABI/testing/sysfs-bus-soundwire-slave @@ -89,3 +89,10 @@ Description: SoundWire Slave Data Source/Sink Port-N DisCo properties. for SoundWire. They define various properties of the Source/Sink Data port N and are used by the bus to configure the Data Port N. + +What: /sys/bus/soundwire/devices/sdw:.../status +Date: September 2020 +Contact: Vinod Koul <vkoul(a)kernel.org> +Description: Soundwire Slave status + This reports the current status of the slave, e.g if it is + UNATTACHED, Attached, Alert or Reserved diff --git a/drivers/soundwire/sysfs_slave.c b/drivers/soundwire/sysfs_slave.c index f510071b0add..3d2cc7612d0d 100644 --- a/drivers/soundwire/sysfs_slave.c +++ b/drivers/soundwire/sysfs_slave.c @@ -97,8 +97,25 @@ static ssize_t modalias_show(struct device *dev, } static DEVICE_ATTR_RO(modalias); +static const char *const slave_status[SDW_SLAVE_MAX] = { + [SDW_SLAVE_UNATTACHED] = "UNATTACHED", + [SDW_SLAVE_ATTACHED] = "Attached", + [SDW_SLAVE_ALERT] = "Alert", + [SDW_SLAVE_RESERVED] = "Reserved", +}; + +static ssize_t status_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + + return sprintf(buf, "%s\n", slave_status[slave->status]); +} +static DEVICE_ATTR_RO(status); + static struct attribute *slave_attrs[] = { &dev_attr_modalias.attr, + &dev_attr_status.attr, NULL, }; ATTRIBUTE_GROUPS(slave); diff --git a/include/linux/soundwire/sdw.h b/include/linux/soundwire/sdw.h index 790823d2d33b..e8c9c20d38b0 100644 --- a/include/linux/soundwire/sdw.h +++ b/include/linux/soundwire/sdw.h @@ -73,12 +73,14 @@ enum { * @SDW_SLAVE_ATTACHED: Slave is attached with bus. * @SDW_SLAVE_ALERT: Some alert condition on the Slave * @SDW_SLAVE_RESERVED: Reserved for future use + * @SDW_SLAVE_MAX: Last status value */ enum sdw_slave_status { SDW_SLAVE_UNATTACHED = 0, SDW_SLAVE_ATTACHED = 1, SDW_SLAVE_ALERT = 2, SDW_SLAVE_RESERVED = 3, + SDW_SLAVE_MAX, }; /** -- 2.26.2

2 1

[PATCH 0/2] ASoC: rt1015p: add rt1015p codec driver
by Tzung-Bi Shih 10 Sep '20

10 Sep '20

Adds rt1015p codec driver. Tzung-Bi Shih (2): ASoC: rt1015p: add codec driver ASoC: dt-bindings: rt1015p: add document .../bindings/sound/realtek,rt1015p.yaml | 36 +++++ sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1015p.c | 148 ++++++++++++++++++ 4 files changed, 193 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml create mode 100644 sound/soc/codecs/rt1015p.c -- 2.28.0.526.ge36021eeef-goog

2 3

[PATCH 0/8] ASoC: q6dsp: Add support to Codec Ports.
by Srinivas Kandagatla 10 Sep '20

10 Sep '20

LPASS IP on SoCs like SM8250 has Digital Codec part integrated into it. This ports are exposed in Q6DSP as Codec ports. This patchset adds support to those q6afe ports along with q6routing and q6afe-dai. This patchset has been tested along with other patches on Qualcomm Robotics RB5 Platform with Soundwire and WSA8815 Codec. Thanks, srini Srinivas Kandagatla (8): ASoC: q6dsp: q6afe: add support to Codec DMA ports ASoC: q6dsp: q6routing: add support to Codec DMA ports ASoC: q6dsp: q6afe: prepare afe_apr_send_pkt to take response opcode ASoC: q6dsp: q6afe: add global q6afe waitqueue ASoC: q6dsp: q6afe: add lpass hw voting support ASoC: q6dsp: q6afe: update q6afe_set_param to support global clocks ASoC: q6dsp: q6afe: add codec lpass clocks ASoC: q6dsp: q6afe-dai: add support to Codec DMA ports include/dt-bindings/sound/qcom,q6afe.h | 22 ++ sound/soc/qcom/qdsp6/q6afe-dai.c | 229 ++++++++++++++++++ sound/soc/qcom/qdsp6/q6afe.c | 308 +++++++++++++++++++++++-- sound/soc/qcom/qdsp6/q6afe.h | 33 ++- sound/soc/qcom/qdsp6/q6routing.c | 121 +++++++++- 5 files changed, 689 insertions(+), 24 deletions(-) -- 2.21.0

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[PATCH 0/2] ASoC: ti: j721e-evm: Support for j7200 variant
by Peter Ujfalusi 10 Sep '20

10 Sep '20

Hi, When j7200 SOM is connected to the CPB, the audio setup is a bit different: Only 48KHz family have clock path, 44.1KHz is not supported. Update the binding documentation and add support for the j7200 version of CPB to the driver. Regards, Peter --- Peter Ujfalusi (2): ASoC: dt-bindings: ti,j721e-cpb-audio: Document support for j7200-cpb ASoC: ti: j721e-evm: Add support for j7200-cpb audio .../bindings/sound/ti,j721e-cpb-audio.yaml | 92 ++++++++++++++----- sound/soc/ti/j721e-evm.c | 11 +++ 2 files changed, 81 insertions(+), 22 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[PATCH v7 0/3] Add documentation and machine driver for SC7180 sound card
by Cheng-Yi Chiang 10 Sep '20

10 Sep '20

Note: - The machine driver patch is made by the collaboration of Cheng-Yi Chiang <cychiang(a)chromium.org> Rohit kumar <rohitkr(a)codeaurora.org> Ajit Pandey <ajitp(a)codeaurora.org> But Ajit has left codeaurora. - This patch series needs HDMI DAI name defined in sc7180-lpass.h. https://patchwork.kernel.org/patch/11745565/ Changes from v1 to v2: - Ducumentation: Addressed all suggestions from Doug. - Machine driver: - Fix comment style for license. - Sort includes. - Remove sc7180_snd_hw_params. - Remove sc7180_dai_init and use aux device instead for headset jack registration. - Statically define format for Primary MI2S. - Atomic is not a concern because there is mutex in card to make sure startup and shutdown happen sequentially. - Fix missing return -EINVAL in startup. - Use static sound card. - Use devm_kzalloc to avoid kfree. Changes from v2 to v3: - Ducumentation: Addressed suggestions from Srini. - Machine driver: - Reuse qcom_snd_parse_of to parse properties. - Remove playback-only and capture-only. - Misc fixes to address comments. Changes from v3 to v4: - Ducumentation: Addressed suggestions from Rob. - Remove definition of dai. - Use 'sound-dai: true' for sound-dai schema. - Add reg property to pass 'make dt_binding_check' check although reg is not used in the driver. - Machine driver: - Add Reviewed-by: Tzung-Bi Shih <tzungbi(a)google.com> Changes from v4 to v5: - Documentation: Addressed suggestions from Rob. - Add definition for "#address-cells" and "#size-cells". - Add additionalProperties: false - Add required properties. Changes from v5 to v6: - Documentation: Addressed suggestions from Rob. - Drop contains in compatible strings. - Only allow dai-link@[0-9] - Remove reg ref since it has a type definition already. Changes from v6 to v7 - Documentation: - Add headset-jack and hdmi-jack to specify the codec responsible for jack detection. - HDMI codec driver: - Use component set_jack ops instead of exporting hdmi_codec_set_jack_detect. - Machine driver: - Removed aux device following Stephan's suggestion. - Use headset-jack and hdmi-jack to specify the codec responsible for jack detection. - Add support for HDMI(actually DP) playback. Ajit Pandey (1): ASoC: qcom: sc7180: Add machine driver for sound card registration Cheng-Yi Chiang (2): ASoC: hdmi-codec: Use set_jack ops to set jack ASoC: qcom: dt-bindings: Add sc7180 machine bindings .../bindings/sound/qcom,sc7180.yaml | 143 +++++++++ include/sound/hdmi-codec.h | 3 - sound/soc/codecs/hdmi-codec.c | 12 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 5 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 5 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 5 +- sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7180.c | 288 ++++++++++++++++++ sound/soc/rockchip/rockchip_max98090.c | 3 +- 10 files changed, 456 insertions(+), 22 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,sc7180.yaml create mode 100644 sound/soc/qcom/sc7180.c -- 2.28.0.526.ge36021eeef-goog

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[PATCH 0/2] soundwire: qcom: fix IP version v1.5.1 support
by Srinivas Kandagatla 10 Sep '20

10 Sep '20

While testing Qualcomm soundwire controller version 1.5.1, found two issue, Firstly the frame shape information configured vs the bus parameters are out of sync. secondly some ports on this ip version require block packing mode support. With this two patches on top of https://lkml.org/lkml/2020/9/5/220 ("[PATCH v2 0/4] soundwire: qcom: add support for mmio soundwire master") patchset I was able to test 2 WSA speakers! This patchset depends on this above patchset! thanks, srini Srinivas Kandagatla (2): soundwire: qcom: add support to block packing mode soundwire: qcom: get max rows and cols info from compatible drivers/soundwire/qcom.c | 68 +++++++++++++++++++++++++++++----------- 1 file changed, 50 insertions(+), 18 deletions(-) -- 2.21.0

2 4

[PATCH -next] soundwire: intel: Remove ununsed function
by YueHaibing 10 Sep '20

10 Sep '20

If CONFIG_PM is not set, build warns: drivers/soundwire/intel.c:488:12: warning: 'intel_link_power_down' defined but not used [-Wunused-function] Move this to #ifdef block. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- drivers/soundwire/intel.c | 137 +++++++++++++++++++------------------- 1 file changed, 68 insertions(+), 69 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index e047910d73f5..ea53291c80dd 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -399,27 +399,6 @@ static void intel_shim_glue_to_master_ip(struct sdw_intel *sdw) /* at this point Master IP has full control of the I/Os */ } -/* this needs to be called with shim_lock */ -static void intel_shim_master_ip_to_glue(struct sdw_intel *sdw) -{ - unsigned int link_id = sdw->instance; - void __iomem *shim = sdw->link_res->shim; - u16 ioctl; - - /* Glue logic */ - ioctl = intel_readw(shim, SDW_SHIM_IOCTL(link_id)); - ioctl |= SDW_SHIM_IOCTL_BKE; - ioctl |= SDW_SHIM_IOCTL_COE; - intel_writew(shim, SDW_SHIM_IOCTL(link_id), ioctl); - usleep_range(10, 15); - - ioctl &= ~(SDW_SHIM_IOCTL_MIF); - intel_writew(shim, SDW_SHIM_IOCTL(link_id), ioctl); - usleep_range(10, 15); - - /* at this point Integration Glue has full control of the I/Os */ -} - static int intel_shim_init(struct sdw_intel *sdw, bool clock_stop) { void __iomem *shim = sdw->link_res->shim; @@ -485,54 +464,6 @@ static void intel_shim_wake(struct sdw_intel *sdw, bool wake_enable) mutex_unlock(sdw->link_res->shim_lock); } -static int intel_link_power_down(struct sdw_intel *sdw) -{ - u32 link_control, spa_mask, cpa_mask; - unsigned int link_id = sdw->instance; - void __iomem *shim = sdw->link_res->shim; - u32 *shim_mask = sdw->link_res->shim_mask; - int ret = 0; - - mutex_lock(sdw->link_res->shim_lock); - - intel_shim_master_ip_to_glue(sdw); - - if (!(*shim_mask & BIT(link_id))) - dev_err(sdw->cdns.dev, - "%s: Unbalanced power-up/down calls\n", __func__); - - *shim_mask &= ~BIT(link_id); - - if (!*shim_mask) { - - dev_dbg(sdw->cdns.dev, "%s: powering down all links\n", __func__); - - /* Link power down sequence */ - link_control = intel_readl(shim, SDW_SHIM_LCTL); - - /* only power-down enabled links */ - spa_mask = FIELD_PREP(SDW_SHIM_LCTL_SPA_MASK, ~sdw->link_res->link_mask); - cpa_mask = FIELD_PREP(SDW_SHIM_LCTL_CPA_MASK, sdw->link_res->link_mask); - - link_control &= spa_mask; - - ret = intel_clear_bit(shim, SDW_SHIM_LCTL, link_control, cpa_mask); - } - - link_control = intel_readl(shim, SDW_SHIM_LCTL); - - mutex_unlock(sdw->link_res->shim_lock); - - if (ret < 0) { - dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); - - return ret; - } - - sdw->cdns.link_up = false; - return 0; -} - static void intel_shim_sync_arm(struct sdw_intel *sdw) { void __iomem *shim = sdw->link_res->shim; @@ -1541,6 +1472,74 @@ int intel_master_process_wakeen_event(struct platform_device *pdev) */ #ifdef CONFIG_PM +/* this needs to be called with shim_lock */ +static void intel_shim_master_ip_to_glue(struct sdw_intel *sdw) +{ + unsigned int link_id = sdw->instance; + void __iomem *shim = sdw->link_res->shim; + u16 ioctl; + + /* Glue logic */ + ioctl = intel_readw(shim, SDW_SHIM_IOCTL(link_id)); + ioctl |= SDW_SHIM_IOCTL_BKE; + ioctl |= SDW_SHIM_IOCTL_COE; + intel_writew(shim, SDW_SHIM_IOCTL(link_id), ioctl); + usleep_range(10, 15); + + ioctl &= ~(SDW_SHIM_IOCTL_MIF); + intel_writew(shim, SDW_SHIM_IOCTL(link_id), ioctl); + usleep_range(10, 15); + + /* at this point Integration Glue has full control of the I/Os */ +} + +static int intel_link_power_down(struct sdw_intel *sdw) +{ + u32 link_control, spa_mask, cpa_mask; + unsigned int link_id = sdw->instance; + void __iomem *shim = sdw->link_res->shim; + u32 *shim_mask = sdw->link_res->shim_mask; + int ret = 0; + + mutex_lock(sdw->link_res->shim_lock); + + intel_shim_master_ip_to_glue(sdw); + + if (!(*shim_mask & BIT(link_id))) + dev_err(sdw->cdns.dev, + "%s: Unbalanced power-up/down calls\n", __func__); + + *shim_mask &= ~BIT(link_id); + + if (!*shim_mask) { + + dev_dbg(sdw->cdns.dev, "%s: powering down all links\n", __func__); + + /* Link power down sequence */ + link_control = intel_readl(shim, SDW_SHIM_LCTL); + + /* only power-down enabled links */ + spa_mask = FIELD_PREP(SDW_SHIM_LCTL_SPA_MASK, ~sdw->link_res->link_mask); + cpa_mask = FIELD_PREP(SDW_SHIM_LCTL_CPA_MASK, sdw->link_res->link_mask); + + link_control &= spa_mask; + + ret = intel_clear_bit(shim, SDW_SHIM_LCTL, link_control, cpa_mask); + } + + link_control = intel_readl(shim, SDW_SHIM_LCTL); + + mutex_unlock(sdw->link_res->shim_lock); + + if (ret < 0) { + dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); + + return ret; + } + + sdw->cdns.link_up = false; + return 0; +} static int __maybe_unused intel_suspend(struct device *dev) { -- 2.17.1

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[PATCH v2 0/7] ASoC/soundwire: filter out invalid PARITY errors
by Bard Liao 10 Sep '20

10 Sep '20

Some codecs may report fake PARITY errors in the initial state. This series will filter them out. Pierre-Louis Bossart (7): ASoC/soundwire: bus: use property to set interrupt masks soundwire: bus: filter-out unwanted interrupt reports soundwire: slave: add first_interrupt_done status soundwire: bus: use quirk to filter out invalid parity errors ASoC: codecs: realtek-soundwire: ignore initial PARITY errors soundwire: bus: export broadcast read/write capability for tests soundwire: cadence: add parity error injection through debugfs drivers/soundwire/bus.c | 93 ++++++++++++++++++++++++------ drivers/soundwire/bus.h | 4 ++ drivers/soundwire/cadence_master.c | 86 +++++++++++++++++++++++++++ drivers/soundwire/slave.c | 1 + include/linux/soundwire/sdw.h | 9 +++ sound/soc/codecs/max98373-sdw.c | 3 + sound/soc/codecs/rt1308-sdw.c | 3 + sound/soc/codecs/rt5682-sdw.c | 5 ++ sound/soc/codecs/rt700-sdw.c | 5 ++ sound/soc/codecs/rt711-sdw.c | 5 ++ sound/soc/codecs/rt715-sdw.c | 5 ++ sound/soc/codecs/wsa881x.c | 1 + 12 files changed, 202 insertions(+), 18 deletions(-) Acked-by: Jaroslav Kysela <perex(a)perex.cz> -- 2.17.1

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[PATCH RESEND 0/2] ASoC: rt1015p: add rt1015p codec driver
by Tzung-Bi Shih 10 Sep '20

10 Sep '20

Adds rt1015p codec driver. Tzung-Bi Shih (2): ASoC: rt1015p: add codec driver ASoC: dt-bindings: rt1015p: add document .../bindings/sound/realtek,rt1015p.yaml | 36 +++++ sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1015p.c | 148 ++++++++++++++++++ 4 files changed, 193 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml create mode 100644 sound/soc/codecs/rt1015p.c -- 2.28.0.526.ge36021eeef-goog

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[PATCH 04/19] ALSA: pcm: Remove unused inline function snd_mask_sizeof
by YueHaibing 09 Sep '20

09 Sep '20

There is no caller in tree, so can remove it. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- include/sound/pcm_params.h | 5 ----- 1 file changed, 5 deletions(-) diff --git a/include/sound/pcm_params.h b/include/sound/pcm_params.h index 36f94735d23d..ba184f49f7e1 100644 --- a/include/sound/pcm_params.h +++ b/include/sound/pcm_params.h @@ -23,11 +23,6 @@ int snd_pcm_hw_param_value(const struct snd_pcm_hw_params *params, #define MASK_OFS(i) ((i) >> 5) #define MASK_BIT(i) (1U << ((i) & 31)) -static inline size_t snd_mask_sizeof(void) -{ - return sizeof(struct snd_mask); -} - static inline void snd_mask_none(struct snd_mask *mask) { memset(mask, 0, sizeof(*mask)); -- 2.17.1

2 1

[PATCH 09/19] ALSA: pci/asihpi: Remove unused function hpi_stream_group_get_map()
by YueHaibing 09 Sep '20

09 Sep '20

There is no caller in tree, so can remove it. Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/pci/asihpi/asihpi.c | 9 --------- 1 file changed, 9 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 35e76480306e..8bacd4f47540 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -258,15 +258,6 @@ static inline u16 hpi_stream_group_reset(u32 h_stream) return hpi_instream_group_reset(h_stream); } -static inline u16 hpi_stream_group_get_map( - u32 h_stream, u32 *mo, u32 *mi) -{ - if (hpi_handle_object(h_stream) == HPI_OBJ_OSTREAM) - return hpi_outstream_group_get_map(h_stream, mo, mi); - else - return hpi_instream_group_get_map(h_stream, mo, mi); -} - static u16 handle_error(u16 err, int line, char *filename) { if (err) -- 2.17.1

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[PATCH v4 1/2] dt-bindings: sound: add DT bindings for Microchip S/PDIF TX Controller
by Codrin Ciubotariu 09 Sep '20

09 Sep '20

This patch adds DT bindings for the new Microchip S/PDIF TX Controller embedded inside sama7g5 SoCs. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changes in v4: - added 'Reviewed-by' tag from Rob Changes in v3: - removed 'oneOf' from 'compatible' property; - added 'maxItems: 1' to 'dmas' property; - removed pinctrl related properties; Changes in v2: - replaced https with http; - reworked example, included bindings; .../bindings/sound/mchp,spdiftx.yaml | 75 +++++++++++++++++++ 1 file changed, 75 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml diff --git a/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml new file mode 100644 index 000000000000..a03b0b871fc9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mchp,spdiftx.yaml @@ -0,0 +1,75 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mchp,spdiftx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Microchip S/PDIF Tx Controller Device Tree Bindings + +maintainers: + - Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> + +description: + The Microchip Sony/Philips Digital Interface Transmitter is a + serial port compliant with the IEC-60958 standard. + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: microchip,sama7g5-spdiftx + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Peripheral Bus Clock + - description: Generic Clock + + clock-names: + items: + - const: pclk + - const: gclk + + dmas: + description: TX DMA Channel + maxItems: 1 + + dma-names: + const: tx + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/at91.h> + #include <dt-bindings/dma/at91.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + + spdiftx@e1618000 { + #sound-dai-cells = <0>; + compatible = "microchip,sama7g5-spdiftx"; + reg = <0xe1618000 0x4000>; + interrupts = <GIC_SPI 85 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dma0 AT91_XDMAC_DT_PERID(50)>; + dma-names = "tx"; + clocks = <&pmc PMC_TYPE_PERIPHERAL 85>, <&pmc PMC_TYPE_GCK 85>; + clock-names = "pclk", "gclk"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_spdiftx_default>; + }; -- 2.25.1

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[RFC] Device addition to the tlv320adcx140 and tas2562
by Dan Murphy 09 Sep '20

09 Sep '20

Hello Looking for guidance on how to move forward with adding a device to existing drivers. For the TLV320adcx140 driver I have another device which is register and bitmap compatible but it does not support Analog microphones or Line In like the x140. Should I add a new driver specifically for this part or should I add the new audio maps, widgets and controls to the existing driver? This would make this existing driver really big in terms of LoC and object size. Similar question for the TAS2562. I have a new device that is register and bitmap compatible but removes the I/V sense. Adding this new device to the existing driver wouldn't be as intrusive as the x140 as the audio map and controls are much simpler Dan

2 1

[PATCH 0/3] ASoC: Kill tasklet usage
by Takashi Iwai 09 Sep '20

09 Sep '20

Hi, this is another series of patches to kill tasklet usages in sound tree, at this time, applied to ASoC drivers. The patches are applied on top the tasklet API conversion patches, found in topic/tasklet-convert branch of sound git tree, which will be included in 5.9-rc4. Takashi === Takashi Iwai (3): ASoC: fsl: Replace tasklet with work ASoC: sh: Replace tasklet with work ASoC: txx9: Replace tasklet with work sound/soc/fsl/fsl_esai.c | 14 +++++++------- sound/soc/sh/siu.h | 2 +- sound/soc/sh/siu_pcm.c | 21 +++++++++++---------- sound/soc/txx9/txx9aclc.c | 11 ++++++----- sound/soc/txx9/txx9aclc.h | 2 +- 5 files changed, 26 insertions(+), 24 deletions(-) -- 2.16.4

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[PATCH] drm/vc4/vc4_hdmi: fill ASoC card owner
by Marek Szyprowski 09 Sep '20

09 Sep '20

card->owner is a required property and since commit 81033c6b584b ("ALSA: core: Warn on empty module") a warning is issued if it is empty. Fix lack of it. This fixes following warning observed on RaspberryPi 3B board with ARM 32bit kernel and multi_v7_defconfig: ------------[ cut here ]------------ WARNING: CPU: 1 PID: 210 at sound/core/init.c:207 snd_card_new+0x378/0x398 [snd] Modules linked in: vc4(+) snd_soc_core ac97_bus snd_pcm_dmaengine bluetooth snd_pcm snd_timer crc32_arm_ce raspberrypi_hwmon snd soundcore ecdh_generic ecc bcm2835_thermal phy_generic CPU: 1 PID: 210 Comm: systemd-udevd Not tainted 5.8.0-rc1-00027-g81033c6b584b #1087 Hardware name: BCM2835 [<c03113c0>] (unwind_backtrace) from [<c030bcb4>] (show_stack+0x10/0x14) [<c030bcb4>] (show_stack) from [<c071cef8>] (dump_stack+0xd4/0xe8) [<c071cef8>] (dump_stack) from [<c0345bfc>] (__warn+0xdc/0xf4) [<c0345bfc>] (__warn) from [<c0345cc4>] (warn_slowpath_fmt+0xb0/0xb8) [<c0345cc4>] (warn_slowpath_fmt) from [<bf02ff74>] (snd_card_new+0x378/0x398 [snd]) [<bf02ff74>] (snd_card_new [snd]) from [<bf11f0b4>] (snd_soc_bind_card+0x280/0x99c [snd_soc_core]) [<bf11f0b4>] (snd_soc_bind_card [snd_soc_core]) from [<bf12f000>] (devm_snd_soc_register_card+0x34/0x6c [snd_soc_core]) [<bf12f000>] (devm_snd_soc_register_card [snd_soc_core]) from [<bf165654>] (vc4_hdmi_bind+0x43c/0x5f4 [vc4]) [<bf165654>] (vc4_hdmi_bind [vc4]) from [<c09d660c>] (component_bind_all+0xec/0x24c) [<c09d660c>] (component_bind_all) from [<bf15c44c>] (vc4_drm_bind+0xd4/0x174 [vc4]) [<bf15c44c>] (vc4_drm_bind [vc4]) from [<c09d6ac0>] (try_to_bring_up_master+0x160/0x1b0) [<c09d6ac0>] (try_to_bring_up_master) from [<c09d6f38>] (component_master_add_with_match+0xd0/0x104) [<c09d6f38>] (component_master_add_with_match) from [<bf15c588>] (vc4_platform_drm_probe+0x9c/0xbc [vc4]) [<bf15c588>] (vc4_platform_drm_probe [vc4]) from [<c09df740>] (platform_drv_probe+0x6c/0xa4) [<c09df740>] (platform_drv_probe) from [<c09dd6f0>] (really_probe+0x210/0x350) [<c09dd6f0>] (really_probe) from [<c09dd940>] (driver_probe_device+0x5c/0xb4) [<c09dd940>] (driver_probe_device) from [<c09ddb38>] (device_driver_attach+0x58/0x60) [<c09ddb38>] (device_driver_attach) from [<c09ddbc0>] (__driver_attach+0x80/0xbc) [<c09ddbc0>] (__driver_attach) from [<c09db820>] (bus_for_each_dev+0x68/0xb4) [<c09db820>] (bus_for_each_dev) from [<c09dc9f8>] (bus_add_driver+0x130/0x1e8) [<c09dc9f8>] (bus_add_driver) from [<c09de648>] (driver_register+0x78/0x110) [<c09de648>] (driver_register) from [<c0302038>] (do_one_initcall+0x50/0x220) [<c0302038>] (do_one_initcall) from [<c03db544>] (do_init_module+0x60/0x210) [<c03db544>] (do_init_module) from [<c03da4f8>] (load_module+0x1e34/0x2338) [<c03da4f8>] (load_module) from [<c03dac00>] (sys_finit_module+0xac/0xbc) [<c03dac00>] (sys_finit_module) from [<c03000c0>] (ret_fast_syscall+0x0/0x54) Exception stack(0xeded9fa8 to 0xeded9ff0) ... ---[ end trace 6414689569c2bc08 ]--- Suggested-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- drivers/gpu/drm/vc4/vc4_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/gpu/drm/vc4/vc4_hdmi.c b/drivers/gpu/drm/vc4/vc4_hdmi.c index 625bfcf52dc4..bdcc54c87d7e 100644 --- a/drivers/gpu/drm/vc4/vc4_hdmi.c +++ b/drivers/gpu/drm/vc4/vc4_hdmi.c @@ -1117,6 +1117,7 @@ static int vc4_hdmi_audio_init(struct vc4_hdmi *hdmi) card->num_links = 1; card->name = "vc4-hdmi"; card->dev = dev; + card->owner = THIS_MODULE; /* * Be careful, snd_soc_register_card() calls dev_set_drvdata() and -- 2.17.1

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Re: [PATCH 1/2] ASoC: tlv320adcx140: Avoid accessing invalid gpio_reset
by Mark Brown 09 Sep '20

09 Sep '20

On Wed, Sep 09, 2020 at 12:06:45PM +0200, Camel Guo wrote: > On 9/8/20 7:20 PM, Mark Brown wrote: > > Applied to > > https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next > I think it is "ASoC: tlv320adcx140: Wake up codec before register setting" > that is applied to for next not this one. As the mail says: > > [1/1] ASoC: tlv320adcx140: Wake up codec before accessing register > > commit: 1a5ce48fd667128e369fdc7fb87e21539aed21b5 You didn't send a cover letter.

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ucm2: USB-Audio: Add support for Lenovo ThinkStation P620 Rear Audio
by GitHub pull_request - opened 09 Sep '20

09 Sep '20

alsa-project/alsa-ucm-conf pull request #50 was opened from khfeng: Add proper PCM assignment to Lenovo ThinkStation P620 Rear Audio. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/50 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/50.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: hda/realtek - The Mic on a RedmiBook doesn't work
by Hui Wang 09 Sep '20

09 Sep '20

The Mic connects to the Nid 0x19, but the configuration of Nid 0x19 is not defined to Mic, and also need to set the coeff to enable the auto detection on the Nid 0x19. After this change, the Mic plugging in or plugging out could be detected and could record the sound from the Mic. And the coeff value is suggested by Kailang of Realtek. Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index abfc602c3b92..85e207173f5d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6228,6 +6228,7 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC255_FIXUP_XIAOMI_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { @@ -7591,6 +7592,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, + { } + }, + .chained = true, + .chain_id = ALC289_FIXUP_ASUS_GA401 + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7888,6 +7899,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), @@ -8065,6 +8077,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, + {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.25.1

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