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

[PATCH v3 1/2] dt-bindings: tas2764: Add the TAS2764 binding doc
by Dan Murphy 09 Oct '20

09 Oct '20

Add the binding for the TAS2764 Smart Amplifier. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2764.yaml | 76 +++++++++++++++++++ 1 file changed, 76 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2764.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2764.yaml b/Documentation/devicetree/bindings/sound/tas2764.yaml new file mode 100644 index 000000000000..5bf8c76ecda1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2764.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2020 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2764.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2764 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2764 is a mono, digital input Class-D audio amplifier optimized for + efficiently driving high peak power into small loudspeakers. + Integrated speaker voltage and current sense provides for + real time monitoring of loudspeaker behavior. + +properties: + compatible: + enum: + - ti,tas2764 + + reg: + maxItems: 1 + description: | + I2C address of the device can be between 0x38 to 0x45. + + reset-gpios: + maxItems: 1 + description: GPIO used to reset the device. + + shutdown-gpios: + maxItems: 1 + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@38 { + compatible = "ti,tas2764"; + reg = <0x38>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpios = <&gpio1 15 0>; + shutdown-gpios = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + +... -- 2.28.0.585.ge1cfff676549

2 2

[PATCH 1/2] ASoC: Intel: catpt: Fix compilation when CONFIG_MODULES is disabled
by Cezary Rojewski 09 Oct '20

09 Oct '20

module_is_live() is available only when CONFIG_MODULES is enabled. Replace its usage with try_module_get() which is present regardless of said config's status. Fixes: 7a10b66a5df9 ("ASoC: Intel: catpt: Device driver lifecycle") Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- sound/soc/intel/catpt/device.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c index 390ffb203de0..a70179959795 100644 --- a/sound/soc/intel/catpt/device.c +++ b/sound/soc/intel/catpt/device.c @@ -81,10 +81,11 @@ static int __maybe_unused catpt_resume(struct device *dev) if (ret) return ret; - if (!module_is_live(dev->driver->owner)) { + if (!try_module_get(dev->driver->owner)) { dev_info(dev, "module unloading, skipping fw boot\n"); return 0; } + module_put(dev->driver->owner); ret = catpt_boot_firmware(cdev, true); if (ret) { @@ -107,10 +108,12 @@ static int __maybe_unused catpt_resume(struct device *dev) static int __maybe_unused catpt_runtime_suspend(struct device *dev) { - if (!module_is_live(dev->driver->owner)) { + if (!try_module_get(dev->driver->owner)) { dev_info(dev, "module unloading, skipping suspend\n"); return 0; } + module_put(dev->driver->owner); + return catpt_suspend(dev); } -- 2.17.1

3 4

[PATCH] ASoC: ti: davinci-mcasp: Use &pdev->dev for early dev_warn
by Peter Ujfalusi 09 Oct '20

09 Oct '20

At this point mcasp->dev is not initialized and we would have NULL pointer dereference if we would have failed to get the mem memory resource by name. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/ti/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 3ffdd0f6292a..a6b72ad53b43 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2155,7 +2155,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!mem) { - dev_warn(mcasp->dev, + dev_warn(&pdev->dev, "\"mpu\" mem resource not found, using index 0\n"); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!mem) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

(Optional?) DMA vs. PIO
by Andy Shevchenko 08 Oct '20

08 Oct '20

Hi! During internal review of one patch I have been puzzled with the following code and Pierre suggested to ask mailing list for help. My main concern is what was the idea behind? Does it mean we support optional DMA in such case? If now, why not to return an error code directly? ---8<---8<---8<--- > Why ASoC core has the following code in the first place: > > 387 chan = dma_request_chan(dev, name); > 388 if (IS_ERR(chan)) { > 389 if (PTR_ERR(chan) == -EPROBE_DEFER) > 390 return -EPROBE_DEFER; > 391 pcm->chan[i] = NULL; > 392 } else { > 393 pcm->chan[i] = chan; > 394 } > > (note lines 389-391). > If PIO fallback is not okay, why not to return an error there? no idea, the code has been this way since 2013 (5eda87b890f867b098e5566b5543642851e8b9c3) It's worth asking the question on the mailing list, I don't know if this is a bug or a feature. -- With Best Regards, Andy Shevchenko

4 3

[PATCH] ALSA: hda/realtek - The front Mic on a HP machine doesn't work
by Jeremy Szu 08 Oct '20

08 Oct '20

On a HP ZCentral, the front Mic could not be detected. The codec of the HP ZCentrol is alc671 and it needs to override the pin configuration to enable the headset mic. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d4f17b465892..add79ed08bdb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9622,6 +9622,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x103c, 0x873e, "HP", ALC671_FIXUP_HP_HEADSET_MIC2), SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x13df, "Asus N550JX", ALC662_FIXUP_BASS_1A), -- 2.28.0

2 1

[PATCH 4/6] ALSA: hdspm: Fix typo arbitary
by Naoki Hayama 08 Oct '20

08 Oct '20

Fix comment typo. s/arbitary/arbitrary/ Signed-off-by: Naoki Hayama <naoki.hayama(a)lineo.co.jp> --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 572350aaf18d..a808b445afd8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1217,7 +1217,7 @@ static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm) return ret; } -/* round arbitary sample rates to commonly known rates */ +/* round arbitrary sample rates to commonly known rates */ static int hdspm_round_frequency(int rate) { if (rate < 38050) -- 2.17.1

2 1

[PATCH 0/6] spelling: Fix typo related to "arbitrary"
by Naoki Hayama 08 Oct '20

08 Oct '20

I found some typos related to "arbitrary". s/abitrary/arbitrary/ s/arbitary/arbitrary/ This series fixes them. These typos have been reported in the past in other codes, but correction 'abitrary||arbitrary' wasn't added to scripts/spelling.txt. Therefore, PATCH #6 adds it to spelling.txt. Naoki Hayama (6): net: tlan: Fix typo abitrary dt-bindings: pinctrl: qcom: Fix typo abitrary dt-bindings: pinctrl: sirf: Fix typo abitrary ALSA: hdspm: Fix typo arbitary drm/etnaviv: Fix typo arbitary scripts/spelling.txt: Add arbitrary correction Documentation/devicetree/bindings/pinctrl/pinctrl-atlas7.txt | 2 +- .../devicetree/bindings/pinctrl/qcom,ipq4019-pinctrl.txt | 2 +- drivers/gpu/drm/etnaviv/etnaviv_gpu.c | 2 +- drivers/net/ethernet/ti/tlan.c | 2 +- scripts/spelling.txt | 1 + sound/pci/rme9652/hdspm.c | 2 +- 6 files changed, 6 insertions(+), 5 deletions(-) -- 2.17.1

1 0

[PATCH 0/6] Ancillary bus implementation and SOF multi-client support
by Dave Ertman 07 Oct '20

07 Oct '20

Brief history of Ancillary Bus ============================== The ancillary bus code was originally submitted upstream as virtual bus, and was submitted through the netdev tree.  This process generated up to v4.  This discussion can be found here: https://lore.kernel.org/netdev/0200520070227.3392100-2-jeffrey.t.kirsher@in… At this point, GregKH requested that we take the review and revision process to an internal mailing list and garner the buy-in of a respected kernel contributor. The ancillary bus (then known as virtual bus) was originally submitted along with implementation code for the ice driver and irdma drive, causing the complication of also having dependencies in the rdma tree. This new submission is utilizing an ancillary bus consumer in only the sound driver tree to create the initial implementation and a single user. Since implementation work has started on this patch set, there have been multiple inquiries about the time frame of its completion. It appears that there will be numerous consumers of this functionality. The process of internal review and implementation using the sound drivers generated 19 internal versions.  The changes, including the name change from virtual bus to ancillary bus, from these versions can be summarized as the following: - Fixed compilation and checkpatch errors - Improved documentation to address the motivation for virtual bus. - Renamed virtual bus to ancillary bus - increased maximum device name size - Correct order in Kconfig and Makefile - removed the mid-layer adev->release layer for device unregister - pushed adev->id management to parent driver - all error paths out of ancillary_device_register return error code - all error paths out of ancillary_device_register use put_device - added adev->name element - modname in register cannot be NULL - added KBUILD_MODNAME as prefix for match_name - push adev->id responsibility to registering driver - uevent now parses adev->dev name - match_id function now parses adev->dev name - changed drivers probe function to also take an ancillary_device_id param - split ancillary_device_register into device_initialize and device_add - adjusted what is done in device_initialize and device_add - change adev to ancildev and adrv to ancildrv - change adev to ancildev in documentation This submission is the first time that this patch set will be sent to the alsa-devel mailing list, so it is currently being submitted as version 1. ========================== Introduces the ancillary bus implementation along with the example usage in the Sound Open Firmware(SOF) audio driver. In some subsystems, the functionality of the core device (PCI/ACPI/other) may be too complex for a single device to be managed as a monolithic block or a part of the functionality might need to be exposed to a different subsystem. Splitting the functionality into smaller orthogonal devices makes it easier to manage data, power management and domain-specific communication with the hardware. Also, common ancillary_device functionality across primary devices can be handled by a common ancillary_device. A key requirement for such a split is that there is no dependency on a physical bus, device, register accesses or regmap support. These individual devices split from the core cannot live on the platform bus as they are not physical devices that are controlled by DT/ACPI. The same argument applies for not using MFD in this scenario as it relies on individual function devices being physical devices that are DT enumerated. An example for this kind of requirement is the audio subsystem where a single IP handles multiple entities such as HDMI, Soundwire, local devices such as mics/speakers etc. The split for the core's functionality can be arbitrary or be defined by the DSP firmware topology and include hooks for test/debug. This allows for the audio core device to be minimal and tightly coupled with handling the hardware-specific logic and communication. The ancillary bus is intended to be minimal, generic and avoid domain-specific assumptions. Each ancillary bus device represents a part of its parent functionality. The generic behavior can be extended and specialized as needed by encapsulating an ancillary bus device within other domain-specific structures and the use of .ops callbacks. The SOF driver adopts the ancillary bus for implementing the multi-client support. A client in the context of the SOF driver represents a part of the core device's functionality. It is not a physical device but rather an ancillary device that needs to communicate with the DSP via IPCs. With multi-client support,the sound card can be separated into multiple orthogonal ancillary devices for local devices (mic/speakers etc), HDMI, sensing, probes, debug etc. In this series, we demonstrate the usage of the ancillary bus with the help of the IPC test client which is used for testing the serialization of IPCs when multiple clients talk to the DSP at the same time. Dave Ertman (1): Add ancillary bus support Fred Oh (1): ASoC: SOF: debug: Remove IPC flood test support in SOF core Ranjani Sridharan (4): ASoC: SOF: Introduce descriptors for SOF client ASoC: SOF: Create client driver for IPC test ASoC: SOF: ops: Add ops for client registration ASoC: SOF: Intel: Define ops for client registration Documentation/driver-api/ancillary_bus.rst | 230 +++++++++++++++ Documentation/driver-api/index.rst | 1 + drivers/bus/Kconfig | 3 + drivers/bus/Makefile | 3 + drivers/bus/ancillary.c | 191 +++++++++++++ include/linux/ancillary_bus.h | 58 ++++ include/linux/mod_devicetable.h | 8 + scripts/mod/devicetable-offsets.c | 3 + scripts/mod/file2alias.c | 8 + sound/soc/sof/Kconfig | 29 +- sound/soc/sof/Makefile | 7 + sound/soc/sof/core.c | 12 + sound/soc/sof/debug.c | 230 --------------- sound/soc/sof/intel/Kconfig | 9 + sound/soc/sof/intel/Makefile | 3 + sound/soc/sof/intel/apl.c | 18 ++ sound/soc/sof/intel/bdw.c | 18 ++ sound/soc/sof/intel/byt.c | 22 ++ sound/soc/sof/intel/cnl.c | 18 ++ sound/soc/sof/intel/intel-client.c | 49 ++++ sound/soc/sof/intel/intel-client.h | 26 ++ sound/soc/sof/ops.h | 14 + sound/soc/sof/sof-client.c | 117 ++++++++ sound/soc/sof/sof-client.h | 65 +++++ sound/soc/sof/sof-ipc-test-client.c | 314 +++++++++++++++++++++ sound/soc/sof/sof-priv.h | 16 +- 26 files changed, 1233 insertions(+), 239 deletions(-) create mode 100644 Documentation/driver-api/ancillary_bus.rst create mode 100644 drivers/bus/ancillary.c create mode 100644 include/linux/ancillary_bus.h create mode 100644 sound/soc/sof/intel/intel-client.c create mode 100644 sound/soc/sof/intel/intel-client.h create mode 100644 sound/soc/sof/sof-client.c create mode 100644 sound/soc/sof/sof-client.h create mode 100644 sound/soc/sof/sof-ipc-test-client.c -- 2.26.2

12 98

[PATCH] ALSA: bebob: potential info leak in hwdep_read()
by Dan Carpenter 07 Oct '20

07 Oct '20

The "count" variable needs to be capped on every path so that we don't copy too much information to the user. Fixes: 618eabeae711 ("ALSA: bebob: Add hwdep interface") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/firewire/bebob/bebob_hwdep.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c index 45b740f44c45..c362eb38ab90 100644 --- a/sound/firewire/bebob/bebob_hwdep.c +++ b/sound/firewire/bebob/bebob_hwdep.c @@ -36,12 +36,11 @@ hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count, } memset(&event, 0, sizeof(event)); + count = min_t(long, count, sizeof(event.lock_status)); if (bebob->dev_lock_changed) { event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS; event.lock_status.status = (bebob->dev_lock_count > 0); bebob->dev_lock_changed = false; - - count = min_t(long, count, sizeof(event.lock_status)); } spin_unlock_irq(&bebob->lock); -- 2.28.0

3 3

[PATCH] ALSA: hda/realtek: Enable audio jacks of ASUS D700SA with ALC887
by Jian-Hong Pan 07 Oct '20

07 Oct '20

The ASUS D700SA desktop's audio (1043:2390) with ALC887 cannot detect the headset microphone and another headphone jack until ALC887_FIXUP_ASUS_HMIC and ALC887_FIXUP_ASUS_AUDIO quirks are applied. The NID 0x15 maps as the headset microphone and NID 0x19 maps as another headphone jack. Also need the function like alc887_fixup_asus_jack to enable the audio jacks. Signed-off-by: Jian-Hong Pan <jhp(a)endlessos.org> Signed-off-by: Kailang Yang <kailang(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 41 +++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d4f17b465892..8d0928bdc9ff 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1929,6 +1929,8 @@ enum { ALC1220_FIXUP_CLEVO_P950, ALC1220_FIXUP_CLEVO_PB51ED, ALC1220_FIXUP_CLEVO_PB51ED_PINS, + ALC887_FIXUP_ASUS_AUDIO, + ALC887_FIXUP_ASUS_HMIC, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2141,6 +2143,30 @@ static void alc1220_fixup_clevo_pb51ed(struct hda_codec *codec, alc_fixup_headset_mode_no_hp_mic(codec, fix, action); } +static void alc887_asus_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + + snd_hda_gen_hp_automute(codec, jack); + + vref = spec->gen.hp_jack_present ? 0xc4 : 0xc0; + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc887_fixup_asus_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + if (action != HDA_FIXUP_ACT_PROBE) + return; + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + 0xc0); + spec->gen.hp_automute_hook = alc887_asus_hp_automute_hook; +} + static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = HDA_FIXUP_PINS, @@ -2398,6 +2424,20 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC1220_FIXUP_CLEVO_PB51ED, }, + [ALC887_FIXUP_ASUS_AUDIO] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x15, 0x02a14150 }, /* use as headset mic, without its own jack detect */ + { 0x19, 0x22219420 }, + {} + }, + }, + [ALC887_FIXUP_ASUS_HMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc887_fixup_asus_jack, + .chained = true, + .chain_id = ALC887_FIXUP_ASUS_AUDIO, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2431,6 +2471,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), + SND_PCI_QUIRK(0x1043, 0x2390, "Asus D700SA", ALC887_FIXUP_ASUS_HMIC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), -- 2.28.0

2 3