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

[PATCH AUTOSEL 5.18 079/159] ASoC: SOF: ipc3-topology: Correct get_control_data for non bytes payload
by Sasha Levin 30 May '22

30 May '22

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit a962890a5a3cce903ff7c7a19fadee63ed9efdc7 ] It is possible to craft a topology where sof_get_control_data() would do out of bounds access because it expects that it is only called when the payload is bytes type. Confusingly it also handles other types of controls, but the payload parsing implementation is only valid for bytes. Fix the code to count the non bytes controls and instead of storing a pointer to sof_abi_hdr in sof_widget_data (which is only valid for bytes), store the pointer to the data itself and add a new member to save the size of the data. In case of non bytes controls we store the pointer to the chanv itself, which is just an array of values at the end. In case of bytes control, drop the wrong cdata->data (wdata[i].pdata) check against NULL since it is incorrect and invalid in this context. The data is pointing to the end of cdata struct, so it should never be null. Reported-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Tested-by: Sergey Senozhatsky <senozhatsky(a)chromium.org> Link: https://lore.kernel.org/r/20220427185221.28928-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/ipc3-topology.c | 39 +++++++++++++++++++++++------------ 1 file changed, 26 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index 2f8450a8c0a1..af1bbd34213c 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -20,7 +20,8 @@ struct sof_widget_data { int ctrl_type; int ipc_cmd; - struct sof_abi_hdr *pdata; + void *pdata; + size_t pdata_size; struct snd_sof_control *control; }; @@ -784,16 +785,26 @@ static int sof_get_control_data(struct snd_soc_component *scomp, } cdata = wdata[i].control->ipc_control_data; - wdata[i].pdata = cdata->data; - if (!wdata[i].pdata) - return -EINVAL; - /* make sure data is valid - data can be updated at runtime */ - if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES && - wdata[i].pdata->magic != SOF_ABI_MAGIC) - return -EINVAL; + if (widget->dobj.widget.kcontrol_type[i] == SND_SOC_TPLG_TYPE_BYTES) { + /* make sure data is valid - data can be updated at runtime */ + if (cdata->data->magic != SOF_ABI_MAGIC) + return -EINVAL; + + wdata[i].pdata = cdata->data->data; + wdata[i].pdata_size = cdata->data->size; + } else { + /* points to the control data union */ + wdata[i].pdata = cdata->chanv; + /* + * wdata[i].control->size is calculated with struct_size + * and includes the size of struct sof_ipc_ctrl_data + */ + wdata[i].pdata_size = wdata[i].control->size - + sizeof(struct sof_ipc_ctrl_data); + } - *size += wdata[i].pdata->size; + *size += wdata[i].pdata_size; /* get data type */ switch (cdata->cmd) { @@ -876,10 +887,12 @@ static int sof_process_load(struct snd_soc_component *scomp, */ if (ipc_data_size) { for (i = 0; i < widget->num_kcontrols; i++) { - memcpy(&process->data[offset], - wdata[i].pdata->data, - wdata[i].pdata->size); - offset += wdata[i].pdata->size; + if (!wdata[i].pdata_size) + continue; + + memcpy(&process->data[offset], wdata[i].pdata, + wdata[i].pdata_size); + offset += wdata[i].pdata_size; } } -- 2.35.1

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[PATCH AUTOSEL 5.18 078/159] ASoC: dapm: Don't fold register value changes into notifications
by Sasha Levin 30 May '22

30 May '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit ad685980469b9f9b99d4d6ea05f4cb8f57cb2234 ] DAPM tracks and reports the value presented to the user from DAPM controls separately to the register value, these may diverge during initialisation or when an autodisable control is in use. When writing DAPM controls we currently report that a change has occurred if either the DAPM value or the value stored in the register has changed, meaning that if the two are out of sync we may appear to report a spurious event to userspace. Since we use this folded in value for nothing other than the value reported to userspace simply drop the folding in of the register change. Signed-off-by: Mark Brown <broonie(a)kernel.org> Link: https://lore.kernel.org/r/20220428161833.3690050-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-dapm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index ca917a849c42..869c76506b66 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3437,7 +3437,6 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mixer_update_power(card, kcontrol, connect, rconnect); @@ -3539,7 +3538,6 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, update.val = val; card->update = &update; } - change |= reg_change; ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); -- 2.35.1

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[PATCH AUTOSEL 5.18 064/159] ASoC: Intel: bytcr_rt5640: Add quirk for the HP Pro Tablet 408
by Sasha Levin 30 May '22

30 May '22

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit ce216cfa84a4e1c23b105e652c550bdeaac9e922 ] Add a quirk for the HP Pro Tablet 408, this BYTCR tablet has no CHAN package in its ACPI tables and uses SSP0-AIF1 rather then SSP0-AIF2 which is the default for BYTCR devices. It also uses DMIC1 for the internal mic rather then the default IN3 and it uses JD2 rather then the default JD1 for jack-detect. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211485 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/20220427134918.527381-1-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 | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index d76a505052fb..f81ae742faa7 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -773,6 +773,18 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_OVCD_SF_0P75 | BYT_RT5640_MCLK_EN), }, + { /* HP Pro Tablet 408 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), + DMI_MATCH(DMI_PRODUCT_NAME, "HP Pro Tablet 408"), + }, + .driver_data = (void *)(BYT_RT5640_DMIC1_MAP | + BYT_RT5640_JD_SRC_JD2_IN4N | + BYT_RT5640_OVCD_TH_1500UA | + BYT_RT5640_OVCD_SF_0P75 | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* HP Stream 7 */ .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "Hewlett-Packard"), -- 2.35.1

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[PATCH AUTOSEL 5.18 045/159] ASoC: rsnd: care return value from rsnd_node_fixed_index()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit d09a7db431c65aaa8303eb456439d1831ca2e6b4 ] 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> Link: https://lore.kernel.org/r/87pmlbgn5t.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/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 6a8fe0da7670..af8ef2a27d34 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); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 044/159] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()
by Sasha Levin 30 May '22

30 May '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit b1384d4c95088d01f4266237faabf165d3d605fc ] commit cfb7b8bf1e2d66 ("ASoC: rsnd: tidyup rsnd_ssiu_busif_err_status_clear()") merged duplicate code, but it didn't care about default case, and causes smatch warnings. smatch warnings: sound/soc/sh/rcar/ssiu.c:112 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'offset'. sound/soc/sh/rcar/ssiu.c:114 rsnd_ssiu_busif_err_status_clear() \ error: uninitialized symbol 'shift'. This patch cares it. 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> Link: https://lore.kernel.org/r/87r15rgn6p.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/sh/rcar/ssiu.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 0d8f97633dd2..138f95dd9f4a 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -102,6 +102,8 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) shift = 1; offset = 1; break; + default: + goto out; } for (i = 0; i < 4; i++) { @@ -120,7 +122,7 @@ bool rsnd_ssiu_busif_err_status_clear(struct rsnd_mod *mod) } rsnd_mod_write(mod, reg, val); } - +out: return error; } -- 2.35.1

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[PATCH AUTOSEL 5.18 028/159] ALSA: jack: Access input_dev under mutex
by Sasha Levin 30 May '22

30 May '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1b6a6fc5280e97559287b61eade2d4b363e836f2 ] It is possible when using ASoC that input_dev is unregistered while calling snd_jack_report, which causes NULL pointer dereference. In order to prevent this serialize access to input_dev using mutex lock. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220412091628.3056922-1-amadeuszx.slawinski@linu… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/jack.h | 1 + sound/core/jack.c | 34 +++++++++++++++++++++++++++------- 2 files changed, 28 insertions(+), 7 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1181f536557e..1ed90e2109e9 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -62,6 +62,7 @@ struct snd_jack { const char *id; #ifdef CONFIG_SND_JACK_INPUT_DEV struct input_dev *input_dev; + struct mutex input_dev_lock; int registered; int type; char name[100]; diff --git a/sound/core/jack.c b/sound/core/jack.c index d1e3055f2b6a..88493cc31914 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -42,8 +42,11 @@ static int snd_jack_dev_disconnect(struct snd_device *device) #ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } /* If the input device is registered with the input subsystem * then we need to use a different deallocator. */ @@ -52,6 +55,7 @@ static int snd_jack_dev_disconnect(struct snd_device *device) else input_free_device(jack->input_dev); jack->input_dev = NULL; + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ return 0; } @@ -90,8 +94,11 @@ static int snd_jack_dev_register(struct snd_device *device) snprintf(jack->name, sizeof(jack->name), "%s %s", card->shortname, jack->id); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return 0; + } jack->input_dev->name = jack->name; @@ -116,6 +123,7 @@ static int snd_jack_dev_register(struct snd_device *device) if (err == 0) jack->registered = 1; + mutex_unlock(&jack->input_dev_lock); return err; } #endif /* CONFIG_SND_JACK_INPUT_DEV */ @@ -517,9 +525,11 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, return -ENOMEM; } - /* don't creat input device for phantom jack */ - if (!phantom_jack) { #ifdef CONFIG_SND_JACK_INPUT_DEV + mutex_init(&jack->input_dev_lock); + + /* don't create input device for phantom jack */ + if (!phantom_jack) { int i; jack->input_dev = input_allocate_device(); @@ -537,8 +547,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -#endif /* CONFIG_SND_JACK_INPUT_DEV */ } +#endif /* CONFIG_SND_JACK_INPUT_DEV */ err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -578,10 +588,14 @@ EXPORT_SYMBOL(snd_jack_new); void snd_jack_set_parent(struct snd_jack *jack, struct device *parent) { WARN_ON(jack->registered); - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } jack->input_dev->dev.parent = parent; + mutex_unlock(&jack->input_dev_lock); } EXPORT_SYMBOL(snd_jack_set_parent); @@ -629,6 +643,8 @@ EXPORT_SYMBOL(snd_jack_set_key); /** * snd_jack_report - Report the current status of a jack + * Note: This function uses mutexes and should be called from a + * context which can sleep (such as a workqueue). * * @jack: The jack to report status for * @status: The current status of the jack @@ -654,8 +670,11 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - if (!jack->input_dev) + mutex_lock(&jack->input_dev_lock); + if (!jack->input_dev) { + mutex_unlock(&jack->input_dev_lock); return; + } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); @@ -675,6 +694,7 @@ void snd_jack_report(struct snd_jack *jack, int status) } input_sync(jack->input_dev); + mutex_unlock(&jack->input_dev_lock); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.1

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[PATCH] ALSA: hda/realtek - Fix microphone noise on ASUS TUF B550M-PLUS
by Marios Levogiannis 30 May '22

30 May '22

Set microphone pins 0x18 (rear) and 0x19 (front) to VREF_50 to fix the microphone noise on ASUS TUF B550M-PLUS which uses the ALCS1200A codec. The initial value was VREF_80. The same issue is also present on Windows using both the default Windows driver and all tested Realtek drivers before version 6.0.9049.1. Comparing Realtek driver 6.0.9049.1 (the first one without the microphone noise) to Realtek driver 6.0.9047.1 (the last one with the microphone noise) revealed that the fix is the result of setting pins 0x18 and 0x19 to VREF_50. This fix may also work for other boards that have been reported to have the same microphone issue and use the ALC1150 and ALCS1200A codecs, since these codecs are similar and the fix in the Realtek driver on Windows is common for both. However, it is currently enabled only for ASUS TUF B550M-PLUS as this is the only board that could be tested. --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 89d610afa..f3ad454b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1981,6 +1981,7 @@ enum { ALC1220_FIXUP_CLEVO_PB51ED_PINS, ALC887_FIXUP_ASUS_AUDIO, ALC887_FIXUP_ASUS_HMIC, + ALCS1200A_FIXUP_MIC_VREF, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -2526,6 +2527,14 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC887_FIXUP_ASUS_AUDIO, }, + [ALCS1200A_FIXUP_MIC_VREF] = { + .type = HDA_FIXUP_PINCTLS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, PIN_VREF50 }, /* rear mic */ + { 0x19, PIN_VREF50 }, /* front mic */ + {} + } + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2563,6 +2572,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { 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), + SND_PCI_QUIRK(0x1043, 0x8797, "ASUS TUF B550M-PLUS", ALCS1200A_FIXUP_MIC_VREF), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9044, "Sony VAIO AiO", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), -- 2.36.1

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[PATCH 00/15] rewrite public API according to convention of GNOME project
by GitHub pull_request - opened 30 May '22

30 May '22

alsa-project/alsa-gobject pull request #70 was opened from takaswie: In GNOME convention for throw function which has an argument of GError to report failure, the function is programmed to return value of gboolean type to report whether the argument is filled with instance of GError. On the other hand, I wrote the throw functions to return nothing (void). This patchset rewrite public API according to it. Unfortunately, it brings loss of backward compatibility. ``` Takashi Sakamoto (15): ctl: remove useless exposed symbols timer: user-instance: fix closure marshaller for event signal ctl: query: rewrite public API to return gboolean according to GNOME convention ctl: card: rewrite public API to return gboolean according to GNOME convention ctl: elem_info: rewrite public API to return gboolean according to GNOME convention rawmidi: query: rewrite public API to return gboolean according to GNOME convention rawmidi: stream_pair: rewrite public API to return gboolean according to GNOME convention hwdep: query: rewrite public API to return gboolean according to GNOME convention timer: query: rewrite public API to return gboolean according to GNOME convention timer: user_instance: rewrite public API to return gboolean according to GNOME convention timer: instance-params: rewrite public API to return gboolean according to GNOME convention seq: query: rewrite public API to return gboolean according to GNOME convention seq: user_client: rewrite public API to return gboolean according to GNOME convention seq: client_info: rewrite public API to return gboolean according to GNOME convention seq: event_cntr: rewrite public API to return gboolean according to GNOME convention samples/ctl | 21 +- samples/hwdep | 8 +- samples/rawmidi | 14 +- samples/seq | 37 +- samples/timer | 9 +- src/ctl/alsactl.map | 73 ++-- src/ctl/card.c | 322 +++++++++------- src/ctl/card.h | 82 ++--- src/ctl/elem-info.c | 84 +++-- src/ctl/elem-info.h | 12 +- src/ctl/query.c | 59 ++- src/ctl/query.h | 8 +- src/hwdep/alsahwdep.map | 13 +- src/hwdep/query.c | 64 ++-- src/hwdep/query.h | 17 +- src/rawmidi/alsarawmidi.map | 37 +- src/rawmidi/privates.h | 2 +- src/rawmidi/query.c | 96 +++-- src/rawmidi/query.h | 37 +- src/rawmidi/stream-pair.c | 163 +++++---- src/rawmidi/stream-pair.h | 31 +- src/seq/alsaseq.map | 98 ++--- src/seq/client-info.c | 38 +- src/seq/client-info.h | 16 +- src/seq/event-cntr.c | 711 +++++++++++++++++++++--------------- src/seq/event-cntr.h | 203 +++++----- src/seq/query.c | 298 +++++++++------ src/seq/query.h | 46 +-- src/seq/user-client.c | 413 +++++++++++++-------- src/seq/user-client.h | 140 +++---- src/timer/alsatimer.map | 44 ++- src/timer/instance-params.c | 30 +- src/timer/instance-params.h | 4 +- src/timer/query.c | 163 ++++++--- src/timer/query.h | 23 +- src/timer/user-instance.c | 192 ++++++---- src/timer/user-instance.h | 43 +-- 37 files changed, 2117 insertions(+), 1534 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/70 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/70.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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[PATCH v2 0/6] OPE support on Tegra210 and later
by Sameer Pujar 30 May '22

30 May '22

This series adds support for Output Prcoessing Module (OPE) module on Tegra210 and later generations of SoCs. OPE is a client of AHUB and it has sub blocks of PEQ (Parametric Equalizer) and MBDRC (Multi Band Dynamic Range Compressor) for data processing. An ASoC component is registered for OPE, which includes PEQ and MBDRC functions as well. This can be plugged in audio path using ALSA mixer controls. The series adds necessary binding documentaion, driver and DT binding patches to enable OPE module on Jetson platforms. Changelog ========= v1 -> v2: --------- * Use generic node names for OPE, PEQ and MBDRC devices. Update binding doc and DT patches for this. * Remove redundant nodename rule enforcement from OPE, PEQ and MBDRC nodes. Update binding doc patch for this. * Fix spaces before binding doc examples and remove '|' from binding doc descriptions. Sameer Pujar (6): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module arm64: defconfig: Build Tegra OPE module arm64: tegra: Add OPE device on Tegra210 and later arm64: tegra: Enable OPE on various platforms .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 47 + .../bindings/sound/nvidia,tegra210-ope.yaml | 87 ++ .../bindings/sound/nvidia,tegra210-peq.yaml | 48 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra186.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 43 + .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 43 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 23 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 84 ++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 40 + .../dts/nvidia/tegra234-p3737-0000+p3701-0000.dts | 43 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 23 + arch/arm64/configs/defconfig | 1 + sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_mbdrc.c | 1012 ++++++++++++++++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 +++++ sound/soc/tegra/tegra210_ope.c | 419 ++++++++ sound/soc/tegra/tegra210_ope.h | 90 ++ sound/soc/tegra/tegra210_peq.c | 434 +++++++++ sound/soc/tegra/tegra210_peq.h | 56 ++ 24 files changed, 2908 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h -- 2.7.4

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ucm2 - splitting 10 channel interface into mix of mono and stereo channels
by GitHub issues - opened 29 May '22

29 May '22

alsa-project/alsa-ucm-conf issue #161 was opened from smangels: Hi, I've pushed a PR for the Behringer Flow8 digital mixer configured to be in "streaming" mode. Now that I've managed to wrap my head around the configuration, I would like to continue with the so called "recording" mode (different ID in `lsusb`). Thus I wonder whether UCM2 syntax allows me to split a 10 channel interface into a mix of mono and stereo channels (mic - mono, master, stereo)? Regards, Sebastian Mangelsen Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/161 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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