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

[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

1 0

[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

1 0

[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

2 3

[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

1 0

[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

2 9

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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[PATCH v2] ASoC: dt-bindings: mediatek: mt8192: Add i2s-share properties
by Nícolas F. R. A. Prado 29 May '22

29 May '22

The Mediatek AFE PCM controller for MT8192 allows two I2S interfaces to share the same clock and act as a single interface with both input and output. Add patterns for these properties in the dt-binding. The property is split into two patterns in order to allow all valid interface pairings. Signed-off-by: Nícolas F. R. A. Prado <nfraprado(a)collabora.com> --- The series from v1 of this patch was merged although some changes were still needed in this patch, so the v1 of this patch was reverted [1] and this standalone commit addresses the feedback from v1 and readds the property. [1] https://lore.kernel.org/all/20220509185625.580811-1-nfraprado@collabora.com v1: https://lore.kernel.org/all/20220429203039.2207848-2-nfraprado@collabora.co… Changes in v2: - Added "mediatek," prefix to property - Rewrote and added more information to property description - Split into two patterns to validate that output-input pairings are done .../bindings/sound/mt8192-afe-pcm.yaml | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml index 7a25bc9b8060..2abf43c6c2c3 100644 --- a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -54,6 +54,22 @@ properties: - const: aud_infra_clk - const: aud_infra_26m_clk +patternProperties: + "^mediatek,i2s[13579]-share$": + description: + Each I2S interface has a single data line, input if its index is even or + output if the index is odd. An input and an output I2S interface can be + used together as if they were a single I2S interface with both input and + output data lines by sharing the same clock. This property represents this + pairing. The value should be the name of the interface whose clock is + used, and the property name the other interface that depends on this + clock. + pattern: "^I2S[0268]$" + + "^mediatek,i2s[0268]-share$": + description: Same as above. + pattern: "^I2S[13579]$" + required: - compatible - interrupts -- 2.36.0

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ucm2 profile for Behringer Flow8
by GitHub pull_request - edited 29 May '22

29 May '22

alsa-project/alsa-ucm-conf pull request #160 was edited from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow8 configured into "streaming" mode - 2 stereo outputs, 1 stereo input, when vendor ID product ID are `1397:050d` - prepare filenames for Flow8's recording mode `1397:050c` (10 chan input, 4 chan output) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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ucm2 profile for Behringer Flow8
by GitHub pull_request - edited 29 May '22

29 May '22

alsa-project/alsa-ucm-conf pull request #160 was edited from smangels: - kudos to Pekka Oinas for supporting me finding the right settings - provides an UCM2 profile for Behringer's digital mixer Flow8 configured into "streaming" mode - 2 stereo outputs, 1 stereo input - prepare filenames for Flow8's recording mode (10 chan input, 4 chan output) Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/160.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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