April 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH 1/2] ASoC: dt-bindings: max98396: add amplifier driver
by Ryan Lee 22 Apr '22

22 Apr '22

From: Ryan Lee <ryans.lee(a)analog.com> This patch adds dt-bindings information for Analog Devices MAX98396 and MAX98397 Smart Amplifier. Signed-off-by: Ryan Lee <ryans.lee(a)analog.com> --- .../bindings/sound/adi,max98396.yaml | 84 +++++++++++++++++++ 1 file changed, 84 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,max98396.yaml diff --git a/Documentation/devicetree/bindings/sound/adi,max98396.yaml b/Documentation/devicetree/bindings/sound/adi,max98396.yaml new file mode 100644 index 000000000000..4aee32f43ad1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,max98396.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/adi,max98396.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Analog Devices MAX98396 Speaker Amplifier Device Tree Bindings + +maintainers: + - Ryan Lee <ryans.lee(a)analog.com> + +description: + The MAX98396 is a mono Class-DG speaker amplifier with I/V sense. + The device provides a PCM interface for audio data and a standard + I2C interface for control data communication. + The MAX98397 is a variant of MAX98396 with wide input supply range. + +properties: + compatible: + enum: + - adi,max98396 + - adi,max98397 + reg: + maxItems: 1 + description: I2C address of the device. + + adi,vmon-slot-no: + description: slot number of the voltage sense monitor + minimum: 0 + maximum: 15 + default: 0 + + adi,imon-slot-no: + description: slot number of the current sense monitor + minimum: 0 + maximum: 15 + default: 0 + + adi,spkfb-slot-no: + description: slot number of speaker DSP monitor + minimum: 0 + maximum: 15 + default: 0 + + adi,interleave-mode: + type: boolean + description: For cases where a single combined channel for the I/V sense data is not sufficient, + the device can also be configured to share a single data output channel on alternating frames. + In this configuration, the current and voltage data will be frame interleaved on a single output channel. + default: false + + reset-gpios: + maxItems: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + max98396: amplifier@39 { + compatible = "adi,max98396"; + reg = <0x39>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + adi,interleave-mode = <0>; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + max98397: amplifier@3c { + compatible = "adi,max98397"; + reg = <0x3c>; + adi,vmon-slot-no = <0>; + adi,imon-slot-no = <1>; + adi,interleave-mode = <0>; + reset-gpios = <&gpio 4 GPIO_ACTIVE_LOW>; + }; + }; + -- 2.25.1

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[PATCH] ALSA: hda: intel-dsp-config: Add RaptorLake PCI IDs
by Pierre-Louis Bossart 21 Apr '22

21 Apr '22

From: Gongjun Song <gongjun.song(a)intel.com> Add RaptorLake-P PCI IDs Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Gongjun Song <gongjun.song(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/intel-dsp-config.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 8b0a16ba27d39..a8fe01764b254 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -424,6 +424,15 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x54c8, }, + /* RaptorLake-P */ + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51ca, + }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51cb, + }, #endif }; -- 2.30.2

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[PATCH] ALSA: hda/realtek: Add quirk for Clevo NP70PNP
by Tim Crawford 21 Apr '22

21 Apr '22

Fixes headset detection on Clevo NP70PNP. Signed-off-by: Tim Crawford <tcrawford(a)system76.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 62fbf3772b41..f0f95e25f3f1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9163,6 +9163,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x866d, "Clevo NP5[05]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x867c, "Clevo NP7[01]PNP", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x867d, "Clevo NP7[01]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME), -- 2.35.1

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[PATCH] ASoC: soc-ops: fix error handling
by Pierre-Louis Bossart 21 Apr '22

21 Apr '22

cppcheck throws the following warning: sound/soc/soc-ops.c:461:8: style: Variable 'ret' is assigned a value that is never used. [unreadVariable] ret = err; ^ This seems to be a missing change in the return value. Fixes: 7f3d90a351968 ("ASoC: ops: Fix stereo change notifications in snd_soc_put_volsw_sx()") Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a0ca58ba16273..58347eadd219b 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -461,7 +461,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, ret = err; } } - return err; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); -- 2.30.2

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[PATCH v2 0/2] ASoC: SOF: AMD updates
by Pierre-Louis Bossart 21 Apr '22

21 Apr '22

these patches were originally submitted in the "ASoC: SOF: AMD/Mediatek updates" series, but there was a bisect issue on the first patch. sound/soc/sof/amd/acp.c:176:29: error: unused variable ‘sdev’ [-Werror=unused-variable] resubmitting as v2 with each patch compiling without warning. Ajit Kumar Pandey (2): ASoC: SOF: amd: Add psp_mbox_ready() and psp_send_cmd() callback ASoC: SOF: amd: Use dedicated MBOX for ACP and PSP communication sound/soc/sof/amd/acp.c | 66 ++++++++++++++++++++++++++++++++++++----- sound/soc/sof/amd/acp.h | 6 ++-- 2 files changed, 63 insertions(+), 9 deletions(-) -- 2.30.2

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[PATCH] ASoC: Intel: sof_ssp_amp: remove 'set-but-not-used' warning
by Pierre-Louis Bossart 21 Apr '22

21 Apr '22

Clang warning: >> sound/soc/intel/boards/sof_ssp_amp.c:97:6: warning: variable 'i' set but not used [-Wunused-but-set-variable] int i = 0; ^ The device counter is not used when the quirk is not set, which static analysis cannot know. Move its initialization before the loop to remove this warning. Reported-by: kernel test robot <lkp(a)intel.com> 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: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/intel/boards/sof_ssp_amp.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_ssp_amp.c b/sound/soc/intel/boards/sof_ssp_amp.c index 02a27aaed92fb..8f5b9ebbe8287 100644 --- a/sound/soc/intel/boards/sof_ssp_amp.c +++ b/sound/soc/intel/boards/sof_ssp_amp.c @@ -94,7 +94,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) char jack_name[NAME_SIZE]; struct sof_hdmi_pcm *pcm; int err; - int i = 0; + int i; if (!(sof_ssp_amp_quirk & SOF_HDMI_PLAYBACK_PRESENT)) return 0; @@ -113,6 +113,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) return hda_dsp_hdmi_build_controls(card, component); } + i = 0; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; snprintf(jack_name, sizeof(jack_name), -- 2.30.2

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[PATCH v1 0/3] ASoC: meson: Fix mixer-test issues
by Mark Brown 21 Apr '22

21 Apr '22

These patches fix event generation issues in the custom controls in the Meson drivers detected by mixer-test and by inspection once I saw the pattern in these drivers. I'm also seeing other failures due to these controls having invalid values, eg: # # AIU ACODEC SRC.0 value 3 more than item count 3 but without documentation I'm not sure how to interpret/fix these - either the value should be fixed up on startup or there should be an extra value there (disconnected possibly?). Mark Brown (3): ASoC: meson: Fix event generation for AUI ACODEC mux ASoC: meson: Fix event generation for AUI CODEC mux ASoC: meson: Fix event generation for G12A tohdmi mux sound/soc/meson/aiu-acodec-ctrl.c | 2 +- sound/soc/meson/aiu-codec-ctrl.c | 2 +- sound/soc/meson/g12a-tohdmitx.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) base-commit: 3123109284176b1532874591f7c81f3837bbdc17 -- 2.30.2

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[V4 1/2] ASoC: max98390: Add reset gpio control
by Steve Lee 21 Apr '22

21 Apr '22

Add reset gpio control to support RESET PIN connected to gpio. Signed-off-by: Steve Lee <steve.lee.analog(a)gmail.com> --- V3 -> V4: change reset gpio control as GPIO_ACTIVE_LOW in DT sound/soc/codecs/max98390.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 40fd6f363f35..bdfd6cccc3d2 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -1022,6 +1022,7 @@ static int max98390_i2c_probe(struct i2c_client *i2c, struct max98390_priv *max98390 = NULL; struct i2c_adapter *adapter = i2c->adapter; + struct gpio_desc *reset_gpio; ret = i2c_check_functionality(adapter, I2C_FUNC_SMBUS_BYTE @@ -1073,6 +1074,17 @@ static int max98390_i2c_probe(struct i2c_client *i2c, return ret; } + reset_gpio = devm_gpiod_get_optional(&i2c->dev, + "reset", GPIOD_OUT_HIGH); + + /* Power on device */ + if (reset_gpio) { + usleep_range(1000, 2000); + /* bring out of reset */ + gpiod_set_value_cansleep(reset_gpio, 0); + usleep_range(1000, 2000); + } + /* Check Revision ID */ ret = regmap_read(max98390->regmap, MAX98390_R24FF_REV_ID, &reg); -- 2.17.1

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[PATCH] ASoC: audio-graph-card2: indicate "Experimental stage" warning only when successed
by Kuninori Morimoto 21 Apr '22

21 Apr '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Because Sound Card needs many drivers to probe, current audio-graph-card2 will indicate "Experimental stage" at top of probe function even though in case it gets -EPROBE_DEFER, thus it will be indicated many times. This patch indicates it when probe was succeeded. [ 1.601393] asoc-audio-graph-card2 sound: Audio Graph Card2 is still under Experimental stage ... [ 1.721269] asoc-audio-graph-card2 sound: Audio Graph Card2 is still under Experimental stage ... [ 1.755231] asoc-audio-graph-card2 sound: Audio Graph Card2 is still under Experimental stage ... [ 1.907710] asoc-audio-graph-card2 sound: Audio Graph Card2 is still under Experimental stage ... [ 1.933173] rcar_sound ec500000.sound: probed [ 1.948875] asoc-audio-graph-card2 sound: Audio Graph Card2 is still under Experimental stage [ 1.959558] asoc-audio-graph-card2 sound: ak4613-hifi <-> rsnd-dai.0 mapping ok [ 1.968119] asoc-audio-graph-card2 sound: i2s-hifi <-> rsnd-dai.1 mapping ok Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card2.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index 75a1aeb75d2c..8b11840d810f 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1174,8 +1174,6 @@ int audio_graph2_parse_of(struct asoc_simple_priv *priv, struct device *dev, struct link_info *li; int ret; - dev_warn(dev, "Audio Graph Card2 is still under Experimental stage\n"); - li = devm_kzalloc(dev, sizeof(*li), GFP_KERNEL); if (!li) return -ENOMEM; @@ -1240,6 +1238,9 @@ int audio_graph2_parse_of(struct asoc_simple_priv *priv, struct device *dev, if (ret < 0) dev_err_probe(dev, ret, "parse error\n"); + if (ret == 0) + dev_warn(dev, "Audio Graph Card2 is still under Experimental stage\n"); + return ret; } EXPORT_SYMBOL_GPL(audio_graph2_parse_of); -- 2.25.1

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[PATCH] ASoC: rsnd: care return value from rsnd_node_fixed_index()
by Kuninori Morimoto 21 Apr '22

21 Apr '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Renesas Sound is very complex, and thus it needs to use rsnd_node_fixed_index() to know enabled pin index. It returns error if strange pin was selected, but some codes didn't check it. This patch 1) indicates error message, 2) check return value. Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 15 ++++++++++----- sound/soc/sh/rcar/dma.c | 9 ++++++++- sound/soc/sh/rcar/rsnd.h | 2 +- sound/soc/sh/rcar/src.c | 7 ++++++- sound/soc/sh/rcar/ssi.c | 14 ++++++++++++-- sound/soc/sh/rcar/ssiu.c | 7 ++++++- 6 files changed, 43 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 0c435bff3670..eb762ab94d3e 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1159,6 +1159,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *np; int i; @@ -1169,7 +1170,11 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + of_node_put(np); + break; + } mod = mod_get(priv, i); @@ -1183,7 +1188,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, of_node_put(node); } -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx) { char node_name[16]; @@ -1210,6 +1215,8 @@ int rsnd_node_fixed_index(struct device_node *node, char *name, int idx) return idx; } + dev_err(dev, "strange node numbering (%s)", + of_node_full_name(node)); return -EINVAL; } @@ -1221,10 +1228,8 @@ int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name i = 0; for_each_child_of_node(node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); if (i < 0) { - dev_err(dev, "strange node numbering (%s)", - of_node_full_name(node)); of_node_put(np); return 0; } diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 03e0d4eca781..463ab237d7bd 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -240,12 +240,19 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, struct rsnd_mod *mod, char *x) { + struct rsnd_priv *priv = rsnd_mod_to_priv(mod); + struct device *dev = rsnd_priv_to_dev(priv); struct dma_chan *chan = NULL; struct device_node *np; int i = 0; for_each_child_of_node(of_node, np) { - i = rsnd_node_fixed_index(np, name, i); + i = rsnd_node_fixed_index(dev, np, name, i); + if (i < 0) { + chan = NULL; + of_node_put(np); + break; + } if (i == rsnd_mod_id_raw(mod) && (!chan)) chan = of_dma_request_slave_channel(np, x); diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 6580bab0e229..d9cd190d7e19 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -460,7 +460,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, char *name, struct device_node *playback, struct device_node *capture); int rsnd_node_count(struct rsnd_priv *priv, struct device_node *node, char *name); -int rsnd_node_fixed_index(struct device_node *node, char *name, int idx); +int rsnd_node_fixed_index(struct device *dev, struct device_node *node, char *name, int idx); int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 42a100c6303d..0ea84ae57c6a 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -676,7 +676,12 @@ int rsnd_src_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SRC_NAME, i); + i = rsnd_node_fixed_index(dev, np, SRC_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_src_probe_done; + } src = rsnd_src_get(priv, i); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 87e606f688d3..43c5e27dc5c8 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -1105,6 +1105,7 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node; struct device_node *np; int i; @@ -1117,7 +1118,11 @@ void rsnd_parse_connect_ssi(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssi_mod_get(priv, i); @@ -1182,7 +1187,12 @@ int rsnd_ssi_probe(struct rsnd_priv *priv) if (!of_device_is_available(np)) goto skip; - i = rsnd_node_fixed_index(np, SSI_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSI_NAME, i); + if (i < 0) { + ret = -EINVAL; + of_node_put(np); + goto rsnd_ssi_probe_done; + } ssi = rsnd_ssi_get(priv, i); diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 138f95dd9f4a..4b8a63e336c7 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -462,6 +462,7 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, struct device_node *capture) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); + struct device *dev = rsnd_priv_to_dev(priv); struct device_node *node = rsnd_ssiu_of_node(priv); struct rsnd_dai_stream *io_p = &rdai->playback; struct rsnd_dai_stream *io_c = &rdai->capture; @@ -474,7 +475,11 @@ void rsnd_parse_connect_ssiu(struct rsnd_dai *rdai, for_each_child_of_node(node, np) { struct rsnd_mod *mod; - i = rsnd_node_fixed_index(np, SSIU_NAME, i); + i = rsnd_node_fixed_index(dev, np, SSIU_NAME, i); + if (i < 0) { + of_node_put(np); + break; + } mod = rsnd_ssiu_mod_get(priv, i); -- 2.25.1

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