May 2021 - Alsa-devel - mailman.alsa-project.org

[PATCH] dt-bindings: sound: wm8750: convert to the json-schema
by Rafał Miłecki 19 May '21

19 May '21

From: Rafał Miłecki <rafal(a)milecki.pl> This helps validating DTS files. Signed-off-by: Rafał Miłecki <rafal(a)milecki.pl> --- .../devicetree/bindings/sound/wm8750.txt | 18 -------- .../devicetree/bindings/sound/wm8750.yaml | 42 +++++++++++++++++++ 2 files changed, 42 insertions(+), 18 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/wm8750.txt create mode 100644 Documentation/devicetree/bindings/sound/wm8750.yaml diff --git a/Documentation/devicetree/bindings/sound/wm8750.txt b/Documentation/devicetree/bindings/sound/wm8750.txt deleted file mode 100644 index 682f221f6f38..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8750.txt +++ /dev/null @@ -1,18 +0,0 @@ -WM8750 and WM8987 audio CODECs - -These devices support both I2C and SPI (configured with pin strapping -on the board). - -Required properties: - - - compatible : "wlf,wm8750" or "wlf,wm8987" - - - reg : the I2C address of the device for I2C, the chip select - number for SPI. - -Example: - -wm8750: codec@1a { - compatible = "wlf,wm8750"; - reg = <0x1a>; -}; diff --git a/Documentation/devicetree/bindings/sound/wm8750.yaml b/Documentation/devicetree/bindings/sound/wm8750.yaml new file mode 100644 index 000000000000..24246ac7bbdf --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8750.yaml @@ -0,0 +1,42 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wm8750.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: WM8750 and WM8987 audio CODECs + +description: | + These devices support both I2C and SPI (configured with pin strapping + on the board). + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + compatible: + enum: + - wlf,wm8750 + - wlf,wm8987 + + reg: + description: + The I2C address of the device for I2C, the chip select number for SPI + maxItems: 1 + +additionalProperties: false + +required: + - reg + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8750"; + reg = <0x1a>; + }; + }; -- 2.26.2

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[PATCH] ASoC: SOF: Intel: hda: don't send DAI_CONFIG IPC for older firmware
by Ranjani Sridharan 19 May '21

19 May '21

BE hw_params op was recently added for SSP type DAIs. But sending the DAI_CONFIG IPC during hw_params is not supported with older firmware. So add an ABI check to avoid sending the IPC if the firmware ABI is older than 3.18. Fixes: e12be9fbfb91 ('ASoC: SOF: Intel: HDA: add hw params callback for SSP DAIs') Tested-by: Yong Zhi <yong.zhi(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 8d7bab433fb3..c1f9f0f58464 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -421,11 +421,16 @@ static int ssp_dai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_component *component = snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; struct sof_ipc_dai_config *config; struct snd_sof_dai *sof_dai; struct sof_ipc_reply reply; int ret; + /* DAI_CONFIG IPC during hw_params is not supported in older firmware */ + if (v->abi_version < SOF_ABI_VER(3, 18, 0)) + return 0; + list_for_each_entry(sof_dai, &sdev->dai_list, list) { if (!sof_dai->cpu_dai_name || !sof_dai->dai_config) continue; -- 2.25.1

2 3

[PATCH] ASoC: meson: g12a-toacodec: add support for SM1 TOACODEC
by Neil Armstrong 19 May '21

19 May '21

This adds support for the TOACODEC found in Amlogic SM1 SoCs. The bits are shifted for more selection of clock sources, so this only maps the same support for G12A to the SM1 bits. Signed-off-by: Neil Armstrong <narmstrong(a)baylibre.com> --- sound/soc/meson/g12a-toacodec.c | 126 +++++++++++++++++++++++++++++++- 1 file changed, 124 insertions(+), 2 deletions(-) diff --git a/sound/soc/meson/g12a-toacodec.c b/sound/soc/meson/g12a-toacodec.c index 9339fabccb79..9c1c6b65baad 100644 --- a/sound/soc/meson/g12a-toacodec.c +++ b/sound/soc/meson/g12a-toacodec.c @@ -23,15 +23,27 @@ #define CTRL0_ENABLE_SHIFT 31 #define CTRL0_DAT_SEL_SHIFT 14 #define CTRL0_DAT_SEL (0x3 << CTRL0_DAT_SEL_SHIFT) +#define CTRL0_DAT_SEL_SHIFT_SM1 18 +#define CTRL0_DAT_SEL_SM1 (0x3 << CTRL0_DAT_SEL_SHIFT_SM1) #define CTRL0_LANE_SEL 12 +#define CTRL0_LANE_SEL_SM1 16 #define CTRL0_LRCLK_SEL GENMASK(9, 8) +#define CTRL0_LRCLK_INV_SM1 BIT(10) +#define CTRL0_LRCLK_SEL_SM1 GENMASK(14, 12) #define CTRL0_BLK_CAP_INV BIT(7) +#define CTRL0_BLK_CAP_INV_SM1 BIT(9) #define CTRL0_BCLK_O_INV BIT(6) +#define CTRL0_BCLK_O_INV_SM1 BIT(8) #define CTRL0_BCLK_SEL GENMASK(5, 4) +#define CTRL0_BCLK_SEL_SM1 GENMASK(6, 4) #define CTRL0_MCLK_SEL GENMASK(2, 0) #define TOACODEC_OUT_CHMAX 2 +struct g12a_toacodec_match_data { + const struct snd_soc_component_driver *component_drv; +}; + static const char * const g12a_toacodec_mux_texts[] = { "I2S A", "I2S B", "I2S C", }; @@ -85,15 +97,73 @@ static int g12a_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol, return 0; } +static int sm1_toacodec_mux_put_enum(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int mux, changed; + + mux = snd_soc_enum_item_to_val(e, ucontrol->value.enumerated.item[0]); + changed = snd_soc_component_test_bits(component, e->reg, + CTRL0_DAT_SEL_SM1, + FIELD_PREP(CTRL0_DAT_SEL_SM1, mux)); + + if (!changed) + return 0; + + /* Force disconnect of the mux while updating */ + snd_soc_dapm_mux_update_power(dapm, kcontrol, 0, NULL, NULL); + + snd_soc_component_update_bits(component, e->reg, + CTRL0_DAT_SEL_SM1 | + CTRL0_LRCLK_SEL_SM1 | + CTRL0_BCLK_SEL_SM1, + FIELD_PREP(CTRL0_DAT_SEL_SM1, mux) | + FIELD_PREP(CTRL0_LRCLK_SEL_SM1, mux) | + FIELD_PREP(CTRL0_BCLK_SEL_SM1, mux)); + + /* + * FIXME: + * On this soc, the glue gets the MCLK directly from the clock + * controller instead of going the through the TDM interface. + * + * Here we assume interface A uses clock A, etc ... While it is + * true for now, it could be different. Instead the glue should + * find out the clock used by the interface and select the same + * source. For that, we will need regmap backed clock mux which + * is a work in progress + */ + snd_soc_component_update_bits(component, e->reg, + CTRL0_MCLK_SEL, + FIELD_PREP(CTRL0_MCLK_SEL, mux)); + + snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); + + return 0; +} + static SOC_ENUM_SINGLE_DECL(g12a_toacodec_mux_enum, TOACODEC_CTRL0, CTRL0_DAT_SEL_SHIFT, g12a_toacodec_mux_texts); +static SOC_ENUM_SINGLE_DECL(sm1_toacodec_mux_enum, TOACODEC_CTRL0, + CTRL0_DAT_SEL_SHIFT_SM1, + g12a_toacodec_mux_texts); + static const struct snd_kcontrol_new g12a_toacodec_mux = SOC_DAPM_ENUM_EXT("Source", g12a_toacodec_mux_enum, snd_soc_dapm_get_enum_double, g12a_toacodec_mux_put_enum); +static const struct snd_kcontrol_new sm1_toacodec_mux = + SOC_DAPM_ENUM_EXT("Source", sm1_toacodec_mux_enum, + snd_soc_dapm_get_enum_double, + sm1_toacodec_mux_put_enum); + static const struct snd_kcontrol_new g12a_toacodec_out_enable = SOC_DAPM_SINGLE_AUTODISABLE("Switch", TOACODEC_CTRL0, CTRL0_ENABLE_SHIFT, 1, 0); @@ -105,6 +175,13 @@ static const struct snd_soc_dapm_widget g12a_toacodec_widgets[] = { &g12a_toacodec_out_enable), }; +static const struct snd_soc_dapm_widget sm1_toacodec_widgets[] = { + SND_SOC_DAPM_MUX("SRC", SND_SOC_NOPM, 0, 0, + &sm1_toacodec_mux), + SND_SOC_DAPM_SWITCH("OUT EN", SND_SOC_NOPM, 0, 0, + &g12a_toacodec_out_enable), +}; + static int g12a_toacodec_input_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -175,6 +252,13 @@ static int g12a_toacodec_component_probe(struct snd_soc_component *c) CTRL0_BLK_CAP_INV); } +static int sm1_toacodec_component_probe(struct snd_soc_component *c) +{ + /* Initialize the static clock parameters */ + return snd_soc_component_write(c, TOACODEC_CTRL0, + CTRL0_BLK_CAP_INV_SM1); +} + static const struct snd_soc_dapm_route g12a_toacodec_routes[] = { { "SRC", "I2S A", "IN A Playback" }, { "SRC", "I2S B", "IN B Playback" }, @@ -187,6 +271,10 @@ static const struct snd_kcontrol_new g12a_toacodec_controls[] = { SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL, 3, 0), }; +static const struct snd_kcontrol_new sm1_toacodec_controls[] = { + SOC_SINGLE("Lane Select", TOACODEC_CTRL0, CTRL0_LANE_SEL_SM1, 3, 0), +}; + static const struct snd_soc_component_driver g12a_toacodec_component_drv = { .probe = g12a_toacodec_component_probe, .controls = g12a_toacodec_controls, @@ -199,25 +287,59 @@ static const struct snd_soc_component_driver g12a_toacodec_component_drv = { .non_legacy_dai_naming = 1, }; +static const struct snd_soc_component_driver sm1_toacodec_component_drv = { + .probe = sm1_toacodec_component_probe, + .controls = sm1_toacodec_controls, + .num_controls = ARRAY_SIZE(sm1_toacodec_controls), + .dapm_widgets = sm1_toacodec_widgets, + .num_dapm_widgets = ARRAY_SIZE(sm1_toacodec_widgets), + .dapm_routes = g12a_toacodec_routes, + .num_dapm_routes = ARRAY_SIZE(g12a_toacodec_routes), + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + static const struct regmap_config g12a_toacodec_regmap_cfg = { .reg_bits = 32, .val_bits = 32, .reg_stride = 4, }; +static const struct g12a_toacodec_match_data g12a_toacodec_match_data = { + .component_drv = &g12a_toacodec_component_drv, +}; + +static const struct g12a_toacodec_match_data sm1_toacodec_match_data = { + .component_drv = &sm1_toacodec_component_drv, +}; + static const struct of_device_id g12a_toacodec_of_match[] = { - { .compatible = "amlogic,g12a-toacodec", }, + { + .compatible = "amlogic,g12a-toacodec", + .data = &g12a_toacodec_match_data, + }, + { + .compatible = "amlogic,sm1-toacodec", + .data = &sm1_toacodec_match_data, + }, {} }; MODULE_DEVICE_TABLE(of, g12a_toacodec_of_match); static int g12a_toacodec_probe(struct platform_device *pdev) { + const struct g12a_toacodec_match_data *data; struct device *dev = &pdev->dev; void __iomem *regs; struct regmap *map; int ret; + data = device_get_match_data(dev); + if (!data) { + dev_err(dev, "failed to match device\n"); + return -ENODEV; + } + ret = device_reset(dev); if (ret) return ret; @@ -234,7 +356,7 @@ static int g12a_toacodec_probe(struct platform_device *pdev) } return devm_snd_soc_register_component(dev, - &g12a_toacodec_component_drv, g12a_toacodec_dai_drv, + data->component_drv, g12a_toacodec_dai_drv, ARRAY_SIZE(g12a_toacodec_dai_drv)); } -- 2.25.1

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[PATCH] ASoC: soc-dai: fix up hw params only if it is needed
by Gyeongtaek Lee 19 May '21

19 May '21

If fixed hw params won't be used, fixing up isn't needed also. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-dai.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 080fbe053fc5..4df1aae8abf3 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -327,14 +327,15 @@ int snd_soc_dai_hw_params(struct snd_soc_dai *dai, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); int ret = 0; - /* perform any topology hw_params fixups before DAI */ - ret = snd_soc_link_be_hw_params_fixup(rtd, params); - if (ret < 0) - goto end; - if (dai->driver->ops && - dai->driver->ops->hw_params) + dai->driver->ops->hw_params) { + /* perform any topology hw_params fixups before DAI */ + ret = snd_soc_link_be_hw_params_fixup(rtd, params); + if (ret < 0) + goto end; + ret = dai->driver->ops->hw_params(substream, params, dai); + } /* mark substream if succeeded */ if (ret == 0) base-commit: c06a2ba62fc401b7aaefd23f5d0bc06d2457ccc1 -- 2.21.0

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[PATCH v11 0/4] ASoC: codecs: add rk817 support
by Chris Morgan 19 May '21

19 May '21

From: Chris Morgan <macromorgan(a)hotmail.com> This series is to add support for the Rockchip rk817 audio codec which is present in all Rockchip rk817 MFD chips. Changes in v11: - Stopped checking for presence of parent node in codec driver, as driver should never be called if parent device doesn't exist to call it. - Made codec subnode optional and stopped returning errors if it is missing. - Downgraded missing codec clock message from dev_err to dev_dbg. Users who choose to use the rk817 PMIC but do not choose to use the codec will not get an error message unless they have debugging enabled, whereas users who wish to use the codec but don't have the clock defined will not get an error message unless they have debug messages enabled. This seems like a fair compromise between mututally exclusive users. Note that errors with the clock or codec will still result in a dev_err. - Updated DT documentation to note that codec subnode is optional if no properties are to exist beneath it. Changes in v10: - Correct order of test/ack/signed-off to chronological order. - Removed ifdef from mfd driver keep cell from being added when the codec was not to be built. - Changed codec DT parsing messages from error to debug to prevent errors in dmesg log in the rare but valid case a user has the rk817 PMIC but does not use the codec. Changes in v9: - Add cover letter. - Remove documentation for interrupt parent per Rob Herring's request. - Remove unused MODULE_DEVICE_TABLE to fix a bug identified by kernel test robot. Changes in v8: - Added additional documentation for missing properties of #sound-dai-cells, interrupt-parent, and wakeup-source for mfd documentation. - Corrected order of elements descriptions in device tree documentation. - Changed name of "mic-in-differential" to "rockchip,mic-in-differential". - Changed name of sound card from "rockchip,rk817-codec" to "Analog". - Removed unused resets and reset-names from the i2s1_2ch node. Changes in v7: - Removed ifdef around register definitions for MFD. - Replaced codec documentation with updates to MFD documentation. - Reordered elements in example to comply with upstream rules. - Added binding update back for Odroid Go Advance as requested. - Submitting patches from gmail now. Changes in v6: - Included additional project maintainers for correct subsystems. - Removed unneeded compatible from DT documentation. - Removed binding update for Odroid Go Advance (will do in separate series). Changes in v5: - Move register definitions from rk817_codec.h to main rk808.h register definitions. - Add volatile register for codec bits. - Add default values for codec bits. - Removed of_compatible from mtd driver (not necessary). - Switched to using parent regmap instead of private regmap for codec. Changes in v4: - Created set_pll() call. - Created user visible gain control in mic. - Check for return value of clk_prepare_enable(). - Removed duplicate clk_prepare_enable(). - Split DT documentation to separate commit. Changes in v3: - Use DAPM macros to set audio path. - Updated devicetree binding (as every rk817 has this codec chip). - Changed documentation to yaml format. - Split MFD changes to separate commit. Changes in v2: - Fixed audio path registers to solve some bugs. Chris Morgan (4): mfd: Add Rockchip rk817 audio CODEC support ASoC: Add Rockchip rk817 audio CODEC support dt-bindings: Add Rockchip rk817 audio CODEC support arm64: dts: rockchip: add rk817 codec to Odroid Go .../devicetree/bindings/mfd/rk808.txt | 188 ++++++ .../boot/dts/rockchip/rk3326-odroid-go2.dts | 36 +- drivers/mfd/rk808.c | 83 +++ include/linux/mfd/rk808.h | 81 +++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 539 ++++++++++++++++++ 7 files changed, 933 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/rk817_codec.c -- 2.25.1

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[PATCH 0/8] ALSA: firewire-lib: check cycle continuity
by Takashi Sakamoto 19 May '21

19 May '21

Hi, Current implementation of ALSA IEC 61883-1/6 packet streaming engine doesn't check whether received packets are exactly per isochronous cycle. This is required to process packets transferred from OXFW970-based devices and devices in RME Fireface series. However, the packet sequence with skipped cycle is inconvenient for media clock recovery. This patchset takes the engine to check cycle continuity at processing packets, including code refactoring. For RME Fireface series, the skipped cycle is handled as receiving an empty packet. For OXFW970-based devices, the skipped cycles are acceptable but media clock recovery is hard. Takashi Sakamoto (8): ALSA: firewire-lib: code refactoring to refer the same frame count per period in domain structure ALSA: firewire-lib: handle the case that empty isochronous packet payload for CIP ALSA: firewire-lib: code refactoring for sequence descriptor' ALSA: firewire-lib: code refactoring for helper function to compute OHCI 1394 cycle ALSA: firewire-lib: code refactoring for parser of IR context header ALSA: firewire-lib: code refactoring for check of CIP header about payload size ALSA: firewire-lib: check cycle continuity ALSA: firewire-lib: insert descriptor for skipped cycle sound/firewire/amdtp-stream.c | 172 ++++++++++++++++++++++------------ sound/firewire/amdtp-stream.h | 10 +- 2 files changed, 119 insertions(+), 63 deletions(-) -- 2.27.0

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[PATCH][next] ALSA: rawmidi: fix incorrect array bounds check on clock_names
by Colin King 19 May '21

19 May '21

From: Colin Ian King <colin.king(a)canonical.com> The array bounds check on clock_names is currently checking the size of the entire array rather than the number of elements in the array leading to a potential array bounds read error. Fix this by using the ARRAY_SIZE macro instead of sizeof. Addresses-Coverity: ("Out-of-bounds read") Fixes: 08fdced60ca0 ("ALSA: rawmidi: Add framing mode") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/core/rawmidi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 4a6534db77d6..6c0a4a67ad2e 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1679,7 +1679,7 @@ static void snd_rawmidi_proc_info_read(struct snd_info_entry *entry, buffer_size, avail, xruns); if (substream->framing == SNDRV_RAWMIDI_MODE_FRAMING_TSTAMP) { clock_type = substream->clock_type >> SNDRV_RAWMIDI_MODE_CLOCK_SHIFT; - if (!snd_BUG_ON(clock_type >= sizeof(clock_names))) + if (!snd_BUG_ON(clock_type >= ARRAY_SIZE(clock_names))) snd_iprintf(buffer, " Framing : tstamp\n" " Clock type : %s\n", -- 2.31.1

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[PATCH 0/2] mfd: wcd934x: add mbhc interrupts
by Srinivas Kandagatla 19 May '21

19 May '21

This patchset adds interrupts required to enable MBHC (Mulit Button Headset Control) support for Jack and headset button detection in WCD938x Codec. Codec side patches are sent in a separate patch set. Tested it on Lenovo Yoga C630. --srini Srinivas Kandagatla (2): mfd: wcd934x: add macro for adding new interrupts mfd: wcd934x: add additional interrupts required for MBHC support drivers/mfd/wcd934x.c | 50 +++++++++++++++++++++---------------------- 1 file changed, 24 insertions(+), 26 deletions(-) -- 2.21.0

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[bug report] ASoC: topology: Add support for multiple kcontrol types to a widget
by Dan Carpenter 19 May '21

19 May '21

Hello Jaska Uimonen, The patch d29d41e28eea: "ASoC: topology: Add support for multiple kcontrol types to a widget" from May 7, 2021, leads to the following static checker warning: sound/soc/soc-topology.c:1559 soc_tplg_dapm_widget_create() error: uninitialized symbol 'kcontrol_type'. sound/soc/soc-topology.c 1418 static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, 1419 struct snd_soc_tplg_dapm_widget *w) 1420 { 1421 struct snd_soc_dapm_context *dapm = &tplg->comp->dapm; 1422 struct snd_soc_dapm_widget template, *widget; 1423 struct snd_soc_tplg_ctl_hdr *control_hdr; 1424 struct snd_soc_card *card = tplg->comp->card; 1425 unsigned int *kcontrol_type; 1426 struct snd_kcontrol_new *kc; 1427 int mixer_count = 0; 1428 int bytes_count = 0; 1429 int enum_count = 0; 1430 int ret = 0; 1431 int i; 1432 1433 if (strnlen(w->name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 1434 SNDRV_CTL_ELEM_ID_NAME_MAXLEN) 1435 return -EINVAL; 1436 if (strnlen(w->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == 1437 SNDRV_CTL_ELEM_ID_NAME_MAXLEN) 1438 return -EINVAL; 1439 1440 dev_dbg(tplg->dev, "ASoC: creating DAPM widget %s id %d\n", 1441 w->name, w->id); 1442 1443 memset(&template, 0, sizeof(template)); 1444 1445 /* map user to kernel widget ID */ 1446 template.id = get_widget_id(le32_to_cpu(w->id)); 1447 if ((int)template.id < 0) 1448 return template.id; 1449 1450 /* strings are allocated here, but used and freed by the widget */ 1451 template.name = kstrdup(w->name, GFP_KERNEL); 1452 if (!template.name) 1453 return -ENOMEM; 1454 template.sname = kstrdup(w->sname, GFP_KERNEL); 1455 if (!template.sname) { 1456 ret = -ENOMEM; 1457 goto err; 1458 } 1459 template.reg = le32_to_cpu(w->reg); 1460 template.shift = le32_to_cpu(w->shift); 1461 template.mask = le32_to_cpu(w->mask); 1462 template.subseq = le32_to_cpu(w->subseq); 1463 template.on_val = w->invert ? 0 : 1; 1464 template.off_val = w->invert ? 1 : 0; 1465 template.ignore_suspend = le32_to_cpu(w->ignore_suspend); 1466 template.event_flags = le16_to_cpu(w->event_flags); 1467 template.dobj.index = tplg->index; 1468 1469 tplg->pos += 1470 (sizeof(struct snd_soc_tplg_dapm_widget) + 1471 le32_to_cpu(w->priv.size)); 1472 1473 if (w->num_kcontrols == 0) { 1474 template.num_kcontrols = 0; 1475 goto widget; This patch changed kcontrol_type to a pointer. Originally if we hit this goto kcontrol_type was zero but now it's uninitialized. This will cause a runtime error with UBSan even if it isn't a real bug. 1476 } 1477 1478 control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; 1479 dev_dbg(tplg->dev, "ASoC: template %s has %d controls of type %x\n", 1480 w->name, w->num_kcontrols, control_hdr->type); 1481 1482 template.num_kcontrols = le32_to_cpu(w->num_kcontrols); 1483 kc = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(*kc), GFP_KERNEL); 1484 if (!kc) 1485 goto err; 1486 1487 kcontrol_type = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(unsigned int), 1488 GFP_KERNEL); 1489 if (!kcontrol_type) 1490 goto err; 1491 1492 for (i = 0; i < w->num_kcontrols; i++) { 1493 control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; 1494 switch (le32_to_cpu(control_hdr->ops.info)) { 1495 case SND_SOC_TPLG_CTL_VOLSW: 1496 case SND_SOC_TPLG_CTL_STROBE: 1497 case SND_SOC_TPLG_CTL_VOLSW_SX: 1498 case SND_SOC_TPLG_CTL_VOLSW_XR_SX: 1499 case SND_SOC_TPLG_CTL_RANGE: 1500 case SND_SOC_TPLG_DAPM_CTL_VOLSW: 1501 /* volume mixer */ 1502 kc[i].index = mixer_count; 1503 kcontrol_type[i] = SND_SOC_TPLG_TYPE_MIXER; 1504 mixer_count++; 1505 ret = soc_tplg_dapm_widget_dmixer_create(tplg, &kc[i]); 1506 if (ret < 0) 1507 goto hdr_err; 1508 break; 1509 case SND_SOC_TPLG_CTL_ENUM: 1510 case SND_SOC_TPLG_CTL_ENUM_VALUE: 1511 case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: 1512 case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: 1513 case SND_SOC_TPLG_DAPM_CTL_ENUM_VALUE: 1514 /* enumerated mixer */ 1515 kc[i].index = enum_count; 1516 kcontrol_type[i] = SND_SOC_TPLG_TYPE_ENUM; 1517 enum_count++; 1518 ret = soc_tplg_dapm_widget_denum_create(tplg, &kc[i]); 1519 if (ret < 0) 1520 goto hdr_err; 1521 break; 1522 case SND_SOC_TPLG_CTL_BYTES: 1523 /* bytes control */ 1524 kc[i].index = bytes_count; 1525 kcontrol_type[i] = SND_SOC_TPLG_TYPE_BYTES; 1526 bytes_count++; 1527 ret = soc_tplg_dapm_widget_dbytes_create(tplg, &kc[i]); 1528 if (ret < 0) 1529 goto hdr_err; 1530 break; 1531 default: 1532 dev_err(tplg->dev, "ASoC: invalid widget control type %d:%d:%d\n", 1533 control_hdr->ops.get, control_hdr->ops.put, 1534 le32_to_cpu(control_hdr->ops.info)); 1535 ret = -EINVAL; 1536 goto hdr_err; 1537 } 1538 } 1539 1540 template.kcontrol_news = kc; 1541 1542 widget: 1543 ret = soc_tplg_widget_load(tplg, &template, w); 1544 if (ret < 0) 1545 goto hdr_err; 1546 1547 /* card dapm mutex is held by the core if we are loading topology 1548 * data during sound card init. */ 1549 if (card->instantiated) 1550 widget = snd_soc_dapm_new_control(dapm, &template); 1551 else 1552 widget = snd_soc_dapm_new_control_unlocked(dapm, &template); 1553 if (IS_ERR(widget)) { 1554 ret = PTR_ERR(widget); 1555 goto hdr_err; 1556 } 1557 1558 widget->dobj.type = SND_SOC_DOBJ_WIDGET; 1559 widget->dobj.widget.kcontrol_type = kcontrol_type; ^^^^^^^^^^^^^^ 1560 widget->dobj.ops = tplg->ops; 1561 widget->dobj.index = tplg->index; 1562 list_add(&widget->dobj.list, &tplg->comp->dobj_list); 1563 1564 ret = soc_tplg_widget_ready(tplg, widget, w); 1565 if (ret < 0) 1566 goto ready_err; regards, dan carpenter

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[PATCH 0/7] ALSA: Prep work for PCI rescan support
by Takashi Iwai 18 May '21

18 May '21

Hi, this is a patch set as a prep work for the future-planned PCI rescan support(*). The essential part is the patch to track in-flight tasks, and the rest are cleanups and fixes in the core code. With those changes, the driver can perform a sort of software suspend on the device without the hardware intervention, which is required during the PCI BAR movement, for example. Takashi (*) https://lore.kernel.org/alsa-devel/e25017c6-e5e4-7a24-e793-14a2e70a434e@amd… === Takashi Iwai (7): ALSA: core: Use READ_ONCE() / WRITE_ONCE() for power state change ALSA: control: Add power state check commonly for ioctl handlers ALSA: control: Drop superfluous snd_power_wait() calls ALSA: control: Minor optimization for SNDRV_CTL_IOCTL_POWER_STATE ALSA: control: Track in-flight control read/write/tlv accesses ALSA: pcm: Block the release until the system resume finishes ALSA: Drop superfluous argument from snd_power_wait() include/sound/core.h | 39 +++++++++++++++++++++--- sound/core/control.c | 41 +++++++++++++------------- sound/core/control_compat.c | 17 +++++------ sound/core/init.c | 59 ++++++++++++++++++++++++++++++++----- sound/core/pcm_native.c | 6 +++- sound/soc/soc-core.c | 2 +- sound/usb/usx2y/us122l.c | 2 +- 7 files changed, 121 insertions(+), 45 deletions(-) -- 2.26.2

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