From 1fd50509fe14a9adc9329e0454b986157a4c155a Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 14 Nov 2024 15:08:07 +0800 Subject: [PATCH 01/16] ALSA: hda/realtek: Update ALC225 depop procedure Old procedure has a chance to meet Headphone no output. Fixes: da911b1f5e98 ("ALSA: hda/realtek - update ALC225 depop optimize") Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/5a27b016ba9d42b4a4e6dadce50a3ba4@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 87 ++++++++++++++++------------------- 1 file changed, 39 insertions(+), 48 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 290c0710f24d..c53a5f8d1559 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3768,33 +3768,28 @@ static void alc225_init(struct hda_codec *codec) hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin); hp2_pin_sense = snd_hda_jack_detect(codec, 0x16); - if (hp1_pin_sense || hp2_pin_sense) + if (hp1_pin_sense || hp2_pin_sense) { msleep(2); + alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */ - alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x1); /* Low power */ - - if (hp1_pin_sense || spec->ultra_low_power) - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - if (hp2_pin_sense) - snd_hda_codec_write(codec, 0x16, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - - if (hp1_pin_sense || hp2_pin_sense || spec->ultra_low_power) - msleep(85); - - if (hp1_pin_sense || spec->ultra_low_power) - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - if (hp2_pin_sense) - snd_hda_codec_write(codec, 0x16, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + if (hp1_pin_sense) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + if (hp2_pin_sense) + snd_hda_codec_write(codec, 0x16, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + msleep(75); - if (hp1_pin_sense || hp2_pin_sense || spec->ultra_low_power) - msleep(100); + if (hp1_pin_sense) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + if (hp2_pin_sense) + snd_hda_codec_write(codec, 0x16, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); - alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */ + msleep(75); + alc_update_coefex_idx(codec, 0x57, 0x04, 0x0007, 0x4); /* Hight power */ + } } static void alc225_shutup(struct hda_codec *codec) @@ -3806,36 +3801,35 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; - alc_disable_headset_jack_key(codec); - /* 3k pull low control for Headset jack. */ - alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); - hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin); hp2_pin_sense = snd_hda_jack_detect(codec, 0x16); - if (hp1_pin_sense || hp2_pin_sense) + if (hp1_pin_sense || hp2_pin_sense) { + alc_disable_headset_jack_key(codec); + /* 3k pull low control for Headset jack. */ + alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); msleep(2); - if (hp1_pin_sense || spec->ultra_low_power) - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - if (hp2_pin_sense) - snd_hda_codec_write(codec, 0x16, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - - if (hp1_pin_sense || hp2_pin_sense || spec->ultra_low_power) - msleep(85); + if (hp1_pin_sense) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + if (hp2_pin_sense) + snd_hda_codec_write(codec, 0x16, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); - if (hp1_pin_sense || spec->ultra_low_power) - snd_hda_codec_write(codec, hp_pin, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); - if (hp2_pin_sense) - snd_hda_codec_write(codec, 0x16, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + msleep(75); - if (hp1_pin_sense || hp2_pin_sense || spec->ultra_low_power) - msleep(100); + if (hp1_pin_sense) + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + if (hp2_pin_sense) + snd_hda_codec_write(codec, 0x16, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + msleep(75); + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); + } alc_auto_setup_eapd(codec, false); alc_shutup_pins(codec); if (spec->ultra_low_power) { @@ -3846,9 +3840,6 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } - - alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); - alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) -- 2.50.1 From 4e7035a75da9371c93dabcb789883e31d2765dcf Mon Sep 17 00:00:00 2001 From: Baojun Xu Date: Sat, 23 Nov 2024 15:37:18 +0800 Subject: [PATCH 02/16] ALSA: hda/tas2781: Add speaker id check for ASUS projects Add speaker id check by gpio in ACPI for ASUS projects. In other vendors, speaker id was checked by BIOS, and was applied in last bit of subsys id, so we can load corresponding firmware binary file for its speaker by subsys id. But in ASUS project, the firmware binary name will be appended an extra number to tell the speakers from different vendors. And this single digit come from gpio level of speaker id in BIOS. Signed-off-by: Baojun Xu Link: https://patch.msgid.link/20241123073718.475-1-baojun.xu@ti.com Signed-off-by: Takashi Iwai --- include/sound/tas2781.h | 1 + sound/pci/hda/tas2781_hda_i2c.c | 63 ++++++++++++++++++++++++++++++--- 2 files changed, 60 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 8cd6da0480b7..72d2060904f6 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -156,6 +156,7 @@ struct tasdevice_priv { struct tasdevice_rca rcabin; struct calidata cali_data; struct tasdevice_fw *fmw; + struct gpio_desc *speaker_id; struct gpio_desc *reset; struct mutex codec_lock; struct regmap *regmap; diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index 370d847517f9..45cfb5a6f309 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -110,10 +111,20 @@ static int tas2781_get_i2c_res(struct acpi_resource *ares, void *data) return 1; } +static const struct acpi_gpio_params speakerid_gpios = { 0, 0, false }; + +static const struct acpi_gpio_mapping tas2781_speaker_id_gpios[] = { + { "speakerid-gpios", &speakerid_gpios, 1 }, + { } +}; + static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) { struct acpi_device *adev; + struct device *physdev; LIST_HEAD(resources); + const char *sub; + uint32_t subid; int ret; adev = acpi_dev_get_first_match_dev(hid, NULL, -1); @@ -123,18 +134,45 @@ static int tas2781_read_acpi(struct tasdevice_priv *p, const char *hid) return -ENODEV; } + physdev = get_device(acpi_get_first_physical_node(adev)); ret = acpi_dev_get_resources(adev, &resources, tas2781_get_i2c_res, p); - if (ret < 0) + if (ret < 0) { + dev_err(p->dev, "Failed to get ACPI resource.\n"); + goto err; + } + sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + if (IS_ERR(sub)) { + dev_err(p->dev, "Failed to get SUBSYS ID.\n"); goto err; + } + /* Speaker id was needed for ASUS projects. */ + ret = kstrtou32(sub, 16, &subid); + if (!ret && upper_16_bits(subid) == PCI_VENDOR_ID_ASUSTEK) { + ret = devm_acpi_dev_add_driver_gpios(p->dev, + tas2781_speaker_id_gpios); + if (ret < 0) + dev_err(p->dev, "Failed to add driver gpio %d.\n", + ret); + p->speaker_id = devm_gpiod_get(p->dev, "speakerid", GPIOD_IN); + if (IS_ERR(p->speaker_id)) { + dev_err(p->dev, "Failed to get Speaker id.\n"); + ret = PTR_ERR(p->speaker_id); + goto err; + } + } else { + p->speaker_id = NULL; + } acpi_dev_free_resource_list(&resources); strscpy(p->dev_name, hid, sizeof(p->dev_name)); + put_device(physdev); acpi_dev_put(adev); return 0; err: dev_err(p->dev, "read acpi error, ret: %d\n", ret); + put_device(physdev); acpi_dev_put(adev); return ret; @@ -615,7 +653,7 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) struct tasdevice_priv *tas_priv = context; struct tas2781_hda *tas_hda = dev_get_drvdata(tas_priv->dev); struct hda_codec *codec = tas_priv->codec; - int i, ret; + int i, ret, spk_id; pm_runtime_get_sync(tas_priv->dev); mutex_lock(&tas_priv->codec_lock); @@ -648,8 +686,25 @@ static void tasdev_fw_ready(const struct firmware *fmw, void *context) tasdevice_dsp_remove(tas_priv); tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; - scnprintf(tas_priv->coef_binaryname, 64, "TAS2XXX%04X.bin", - codec->core.subsystem_id & 0xffff); + if (tas_priv->speaker_id != NULL) { + // Speaker id need to be checked for ASUS only. + spk_id = gpiod_get_value(tas_priv->speaker_id); + if (spk_id < 0) { + // Speaker id is not valid, use default. + dev_dbg(tas_priv->dev, "Wrong spk_id = %d\n", spk_id); + spk_id = 0; + } + snprintf(tas_priv->coef_binaryname, + sizeof(tas_priv->coef_binaryname), + "TAS2XXX%04X%d.bin", + lower_16_bits(codec->core.subsystem_id), + spk_id); + } else { + snprintf(tas_priv->coef_binaryname, + sizeof(tas_priv->coef_binaryname), + "TAS2XXX%04X.bin", + lower_16_bits(codec->core.subsystem_id)); + } ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", -- 2.50.1 From 155699ccab7c78cbba69798242b68bc8ac66d5d2 Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 21 Nov 2024 16:16:26 +0800 Subject: [PATCH 03/16] ALSA: hda/realtek: Set PCBeep to default value for ALC274 BIOS Enable PC beep path cause pop noise via speaker during boot time. Set to default value from driver will solve the issue. Signed-off-by: Kailang Yang Cc: Link: https://lore.kernel.org/2721bb57e20a44c3826c473e933f9105@realtek.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c53a5f8d1559..d950666f9c74 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -473,6 +473,8 @@ static void alc_fill_eapd_coef(struct hda_codec *codec) break; case 0x10ec0234: case 0x10ec0274: + alc_write_coef_idx(codec, 0x6e, 0x0c25); + fallthrough; case 0x10ec0294: case 0x10ec0700: case 0x10ec0701: -- 2.50.1 From a166f80343cd436d6d414199d18ad0ab291caaa5 Mon Sep 17 00:00:00 2001 From: Zhu Jun Date: Tue, 26 Nov 2024 01:32:45 -0800 Subject: [PATCH 04/16] ALSA: asihpi: Remove unused variable the variable is never referenced in the code, just remove it. Signed-off-by: Zhu Jun Link: https://patch.msgid.link/20241126093245.3228-1-zhujun2@cmss.chinamobile.com Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index fdd4fe16225f..5a84591b13fc 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -464,7 +464,7 @@ static int snd_card_asihpi_pcm_hw_params(struct snd_pcm_substream *substream, return -ENOMEM; } - err = hpi_stream_get_info_ex(dpcm->h_stream, NULL, + hpi_stream_get_info_ex(dpcm->h_stream, NULL, &dpcm->hpi_buffer_attached, NULL, NULL, NULL); } bytes_per_sec = params_rate(params) * params_channels(params); -- 2.50.1 From db2eee61434808d66233a9d3ea5ec31b8867de23 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 26 Nov 2024 15:10:10 +0100 Subject: [PATCH 05/16] ALSA: hda: Show the codec quirk info at probing Lots of HD-audio devices need the device-specific quirk, and it's often helpful to know which quirk is applied. But currently the driver shows it only as a debug output, hence we'd have to enable the debug option at each time we want to see it (and the output becomes too messy due to other debug messages). This patch changes the messages to the info level, so that they appear at probing normally. Link: https://patch.msgid.link/20241126141010.12567-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 8e74be038b0f..84393f4f429d 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -933,6 +933,7 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, bool match_all_pins) { const struct snd_hda_pin_quirk *pq; + const char *name = NULL; if (codec->fixup_id != HDA_FIXUP_ID_NOT_SET) return; @@ -946,9 +947,10 @@ void snd_hda_pick_pin_fixup(struct hda_codec *codec, codec->fixup_id = pq->value; #ifdef CONFIG_SND_DEBUG_VERBOSE codec->fixup_name = pq->name; - codec_dbg(codec, "%s: picked fixup %s (pin match)\n", - codec->core.chip_name, codec->fixup_name); + name = pq->name; #endif + codec_info(codec, "%s: picked fixup %s (pin match)\n", + codec->core.chip_name, name ? name : ""); codec->fixup_list = fixlist; return; } @@ -1015,8 +1017,8 @@ void snd_hda_pick_fixup(struct hda_codec *codec, if (codec->modelname && !strcmp(codec->modelname, "nofixup")) { id = HDA_FIXUP_ID_NO_FIXUP; fixlist = NULL; - codec_dbg(codec, "%s: picked no fixup (nofixup specified)\n", - codec->core.chip_name); + codec_info(codec, "%s: picked no fixup (nofixup specified)\n", + codec->core.chip_name); goto found; } @@ -1026,8 +1028,8 @@ void snd_hda_pick_fixup(struct hda_codec *codec, if (!strcmp(codec->modelname, models->name)) { id = models->id; name = models->name; - codec_dbg(codec, "%s: picked fixup %s (model specified)\n", - codec->core.chip_name, codec->fixup_name); + codec_info(codec, "%s: picked fixup %s (model specified)\n", + codec->core.chip_name, name); goto found; } models++; @@ -1085,9 +1087,9 @@ void snd_hda_pick_fixup(struct hda_codec *codec, #ifdef CONFIG_SND_DEBUG_VERBOSE name = q->name; #endif - codec_dbg(codec, "%s: picked fixup %s for %s %04x:%04x\n", - codec->core.chip_name, name ? name : "", - type, q->subvendor, q->subdevice); + codec_info(codec, "%s: picked fixup %s for %s %04x:%04x\n", + codec->core.chip_name, name ? name : "", + type, q->subvendor, q->subdevice); found: codec->fixup_id = id; codec->fixup_list = fixlist; -- 2.50.1 From 9ad467a2b2716d4ed12f003b041aa6c776a13ff5 Mon Sep 17 00:00:00 2001 From: Zichen Xie Date: Tue, 26 Nov 2024 13:24:49 -0600 Subject: [PATCH 06/16] ALSA: core: Fix possible NULL dereference caused by kunit_kzalloc() kunit_kzalloc() may return a NULL pointer, dereferencing it without NULL check may lead to NULL dereference. Add NULL checks for all the kunit_kzalloc() in sound_kunit.c Fixes: 3e39acf56ede ("ALSA: core: Add sound core KUnit test") Signed-off-by: Zichen Xie Link: https://patch.msgid.link/20241126192448.12645-1-zichenxie0106@gmail.com Signed-off-by: Takashi Iwai --- sound/core/sound_kunit.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/core/sound_kunit.c b/sound/core/sound_kunit.c index bfed1a25fc8f..84e337ecbddd 100644 --- a/sound/core/sound_kunit.c +++ b/sound/core/sound_kunit.c @@ -172,6 +172,7 @@ static void test_format_fill_silence(struct kunit *test) u32 i, j; buffer = kunit_kzalloc(test, SILENCE_BUFFER_SIZE, GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, buffer); for (i = 0; i < ARRAY_SIZE(buf_samples); i++) { for (j = 0; j < ARRAY_SIZE(valid_fmt); j++) @@ -208,8 +209,12 @@ static void test_playback_avail(struct kunit *test) struct snd_pcm_runtime *r = kunit_kzalloc(test, sizeof(*r), GFP_KERNEL); u32 i; + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, r); + r->status = kunit_kzalloc(test, sizeof(*r->status), GFP_KERNEL); r->control = kunit_kzalloc(test, sizeof(*r->control), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, r->status); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, r->control); for (i = 0; i < ARRAY_SIZE(p_avail_data); i++) { r->buffer_size = p_avail_data[i].buffer_size; @@ -232,8 +237,12 @@ static void test_capture_avail(struct kunit *test) struct snd_pcm_runtime *r = kunit_kzalloc(test, sizeof(*r), GFP_KERNEL); u32 i; + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, r); + r->status = kunit_kzalloc(test, sizeof(*r->status), GFP_KERNEL); r->control = kunit_kzalloc(test, sizeof(*r->control), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, r->status); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, r->control); for (i = 0; i < ARRAY_SIZE(c_avail_data); i++) { r->buffer_size = c_avail_data[i].buffer_size; @@ -247,6 +256,7 @@ static void test_capture_avail(struct kunit *test) static void test_card_set_id(struct kunit *test) { struct snd_card *card = kunit_kzalloc(test, sizeof(*card), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, card); snd_card_set_id(card, VALID_NAME); KUNIT_EXPECT_STREQ(test, card->id, VALID_NAME); @@ -280,6 +290,7 @@ static void test_pcm_format_name(struct kunit *test) static void test_card_add_component(struct kunit *test) { struct snd_card *card = kunit_kzalloc(test, sizeof(*card), GFP_KERNEL); + KUNIT_ASSERT_NOT_ERR_OR_NULL(test, card); snd_component_add(card, TEST_FIRST_COMPONENT); KUNIT_ASSERT_STREQ(test, card->components, TEST_FIRST_COMPONENT); -- 2.50.1 From 7be34f6feedd60e418de1c2c48e661d70416635f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Nov 2024 08:00:58 +0100 Subject: [PATCH 07/16] ALSA: ump: Fix evaluation of MIDI 1.0 FB info The m1.0 field of UMP Function Block info specifies whether the given FB is a MIDI 1.0 port or not. When implementing the UMP support on Linux, I somehow interpreted as if it were bit flags, but the field is actually an enumeration from 0 to 2, where 2 means MIDI 1.0 *and* low speed. This patch corrects the interpretation and sets the right bit flags depending on the m1.0 field of FB Info. This effectively fixes the missing detection of MIDI 1.0 FB when m1.0 is 2. Fixes: 37e0e14128e0 ("ALSA: ump: Support UMP Endpoint and Function Block parsing") Cc: Link: https://patch.msgid.link/20241127070059.8099-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/ump.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/ump.c b/sound/core/ump.c index 5d4dd207e5ab..6d0aac6c763f 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -788,7 +788,10 @@ static void fill_fb_info(struct snd_ump_endpoint *ump, info->ui_hint = buf->fb_info.ui_hint; info->first_group = buf->fb_info.first_group; info->num_groups = buf->fb_info.num_groups; - info->flags = buf->fb_info.midi_10; + if (buf->fb_info.midi_10 < 2) + info->flags = buf->fb_info.midi_10; + else + info->flags = SNDRV_UMP_BLOCK_IS_MIDI1 | SNDRV_UMP_BLOCK_IS_LOWSPEED; info->active = buf->fb_info.active; info->midi_ci_version = buf->fb_info.midi_ci_version; info->sysex8_streams = buf->fb_info.sysex8_streams; -- 2.50.1 From 4095cf872084ecfdfdb0e681f3e9ff9745acfa75 Mon Sep 17 00:00:00 2001 From: Venkata Prasad Potturu Date: Wed, 27 Nov 2024 16:52:25 +0530 Subject: [PATCH 08/16] ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry Add condition check to register ACP PDM sound card by reading _WOV acpi entry. Fixes: 5426f506b584 ("ASoC: amd: Add support for enabling DMIC on acp6x via _DSD") Signed-off-by: Venkata Prasad Potturu Link: https://patch.msgid.link/20241127112227.227106-1-venkataprasad.potturu@amd.com Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index 6439c175552a..facd82f0f251 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -551,8 +551,14 @@ static int acp6x_probe(struct platform_device *pdev) struct acp6x_pdm *machine = NULL; struct snd_soc_card *card; struct acpi_device *adev; + acpi_handle handle; + acpi_integer dmic_status; int ret; + bool is_dmic_enable, wov_en; + /* IF WOV entry not found, enable dmic based on AcpDmicConnected entry*/ + is_dmic_enable = false; + wov_en = true; /* check the parent device's firmware node has _DSD or not */ adev = ACPI_COMPANION(pdev->dev.parent); if (adev) { @@ -560,9 +566,19 @@ static int acp6x_probe(struct platform_device *pdev) if (!acpi_dev_get_property(adev, "AcpDmicConnected", ACPI_TYPE_INTEGER, &obj) && obj->integer.value == 1) - platform_set_drvdata(pdev, &acp6x_card); + is_dmic_enable = true; } + handle = ACPI_HANDLE(pdev->dev.parent); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (!ACPI_FAILURE(ret)) + wov_en = dmic_status; + + if (is_dmic_enable && wov_en) + platform_set_drvdata(pdev, &acp6x_card); + else + return 0; + /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.50.1 From 2f2020327cc8561d7c520d2f2d9acea84fa7b3a3 Mon Sep 17 00:00:00 2001 From: =?utf8?q?N=C3=ADcolas=20F=2E=20R=2E=20A=2E=20Prado?= Date: Tue, 26 Nov 2024 15:09:43 -0500 Subject: [PATCH 09/16] ASoC: mediatek: Check num_codecs is not zero to avoid panic during probe MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit Following commit 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()"), COMP_DUMMY() became an array with zero length, and only gets populated with the dummy struct after the card is registered. Since the sound card driver's probe happens before the card registration, accessing any of the members of a dummy component during probe will result in undefined behavior. This can be observed in the mt8188 and mt8195 machine sound drivers. By omitting a dai link subnode in the sound card's node in the Devicetree, the default uninitialized dummy codec is used, and when its dai_name pointer gets passed to strcmp() it results in a null pointer dereference and a kernel panic. In addition to that, set_card_codec_info() in the generic helpers file, mtk-soundcard-driver.c, will populate a dai link with a dummy codec when a dai link node is present in DT but with no codec property. The result is that at probe time, a dummy codec can either be uninitialized with num_codecs = 0, or be an initialized dummy codec, with num_codecs = 1 and dai_name = "snd-soc-dummy-dai". In order to accommodate for both situations, check that num_codecs is not zero before accessing the codecs' fields but still check for the codec's dai name against "snd-soc-dummy-dai" as needed. While at it, also drop the check that dai_name is not null in the mt8192 driver, introduced in commit 4d4e1b6319e5 ("ASoC: mediatek: mt8192: Check existence of dai_name before dereferencing"), as it is actually redundant given the preceding num_codecs != 0 check. Fixes: 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()") Signed-off-by: Nícolas F. R. A. Prado Reviewed-by: AngeloGioacchino Del Regno Acked-by: Kuninori Morimoto Reviewed-by: Fei Shao Acked-by: Trevor Wu Link: https://patch.msgid.link/20241126-asoc-mtk-dummy-panic-v1-1-42d53e168d2e@collabora.com Signed-off-by: Mark Brown --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 9 +++++++-- sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 ++-- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 9 +++++++-- 3 files changed, 16 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index 84abdba9ddb6..e04b88a57535 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -1277,10 +1277,12 @@ static int mt8188_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, for_each_card_prelinks(card, i, dai_link) { if (strcmp(dai_link->name, "DPTX_BE") == 0) { - if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + if (dai_link->num_codecs && + strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) dai_link->init = mt8188_dptx_codec_init; } else if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { - if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + if (dai_link->num_codecs && + strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) dai_link->init = mt8188_hdmi_codec_init; } else if (strcmp(dai_link->name, "DL_SRC_BE") == 0 || strcmp(dai_link->name, "UL_SRC_BE") == 0) { @@ -1292,6 +1294,9 @@ static int mt8188_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, strcmp(dai_link->name, "ETDM2_OUT_BE") == 0 || strcmp(dai_link->name, "ETDM1_IN_BE") == 0 || strcmp(dai_link->name, "ETDM2_IN_BE") == 0) { + if (!dai_link->num_codecs) + continue; + if (!strcmp(dai_link->codecs->dai_name, MAX98390_CODEC_DAI)) { /* * The TDM protocol settings with fixed 4 slots are defined in diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 1aba9c75594e..b1598cc5587e 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -1091,7 +1091,7 @@ static int mt8192_mt6359_legacy_probe(struct mtk_soc_card_data *soc_card_data) dai_link->ignore = 0; } - if (dai_link->num_codecs && dai_link->codecs[0].dai_name && + if (dai_link->num_codecs && strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0) dai_link->ops = &mt8192_rt1015_i2s_ops; } @@ -1119,7 +1119,7 @@ static int mt8192_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, int i; for_each_card_prelinks(card, i, dai_link) - if (dai_link->num_codecs && dai_link->codecs[0].dai_name && + if (dai_link->num_codecs && strcmp(dai_link->codecs[0].dai_name, RT1015_CODEC_DAI) == 0) dai_link->ops = &mt8192_rt1015_i2s_ops; } diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359.c b/sound/soc/mediatek/mt8195/mt8195-mt6359.c index 56b9d2433a1e..2b9cb3248795 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359.c @@ -1378,10 +1378,12 @@ static int mt8195_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, for_each_card_prelinks(card, i, dai_link) { if (strcmp(dai_link->name, "DPTX_BE") == 0) { - if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + if (dai_link->num_codecs && + strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) dai_link->init = mt8195_dptx_codec_init; } else if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { - if (strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) + if (dai_link->num_codecs && + strcmp(dai_link->codecs->dai_name, "snd-soc-dummy-dai")) dai_link->init = mt8195_hdmi_codec_init; } else if (strcmp(dai_link->name, "DL_SRC_BE") == 0 || strcmp(dai_link->name, "UL_SRC1_BE") == 0 || @@ -1394,6 +1396,9 @@ static int mt8195_mt6359_soc_card_probe(struct mtk_soc_card_data *soc_card_data, strcmp(dai_link->name, "ETDM2_OUT_BE") == 0 || strcmp(dai_link->name, "ETDM1_IN_BE") == 0 || strcmp(dai_link->name, "ETDM2_IN_BE") == 0) { + if (!dai_link->num_codecs) + continue; + if (!strcmp(dai_link->codecs->dai_name, MAX98390_CODEC_DAI)) { if (!(codec_init & MAX98390_CODEC_INIT)) { dai_link->init = mt8195_max98390_init; -- 2.50.1 From e9db1b551774037ebe39dde4a658d89ba95e260b Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Wed, 27 Nov 2024 17:29:54 +0800 Subject: [PATCH 10/16] ASoC: SOF: ipc3-topology: Convert the topology pin index to ALH dai index MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit Intel SoundWire machine driver always uses Pin number 2 and above. Currently, the pin number is used as the FW DAI index directly. As a result, FW DAI 0 and 1 are never used. That worked fine because we use up to 2 DAIs in a SDW link. Convert the topology pin index to ALH dai index, the mapping is using 2-off indexing, iow, pin #2 is ALH dai #0. The issue exists since beginning. And the Fixes tag is the first commit that this commit can be applied. Fixes: b66bfc3a9810 ("ASoC: SOF: sof-audio: Fix broken early bclk feature for SSP") Signed-off-by: Bard Liao Reviewed-by: Péter Ujfalusi Reviewed-by: Liam Girdwood Reviewed-by: Kai Vehmanen Reviewed-by: Ranjani Sridharan Link: https://patch.msgid.link/20241127092955.20026-1-yung-chuan.liao@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc3-topology.c | 26 ++++++++++++++++++++++++-- 1 file changed, 24 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index be61e377e59e..c2fce554a674 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -20,6 +20,9 @@ /* size of tplg ABI in bytes */ #define SOF_IPC3_TPLG_ABI_SIZE 3 +/* Base of SOF_DAI_INTEL_ALH, this should be aligned with SOC_SDW_INTEL_BIDIR_PDI_BASE */ +#define INTEL_ALH_DAI_INDEX_BASE 2 + struct sof_widget_data { int ctrl_type; int ipc_cmd; @@ -1594,6 +1597,17 @@ static int sof_ipc3_widget_setup_comp_dai(struct snd_sof_widget *swidget) if (ret < 0) goto free; + /* Subtract the base to match the FW dai index. */ + if (comp_dai->type == SOF_DAI_INTEL_ALH) { + if (comp_dai->dai_index < INTEL_ALH_DAI_INDEX_BASE) { + dev_err(sdev->dev, + "Invalid ALH dai index %d, only Pin numbers >= %d can be used\n", + comp_dai->dai_index, INTEL_ALH_DAI_INDEX_BASE); + return -EINVAL; + } + comp_dai->dai_index -= INTEL_ALH_DAI_INDEX_BASE; + } + dev_dbg(scomp->dev, "dai %s: type %d index %d\n", swidget->widget->name, comp_dai->type, comp_dai->dai_index); sof_dbg_comp_config(scomp, &comp_dai->config); @@ -2167,8 +2181,16 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * case SOF_DAI_INTEL_ALH: if (data) { /* save the dai_index during hw_params and reuse it for hw_free */ - if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) - config->dai_index = data->dai_index; + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + /* Subtract the base to match the FW dai index. */ + if (data->dai_index < INTEL_ALH_DAI_INDEX_BASE) { + dev_err(sdev->dev, + "Invalid ALH dai index %d, only Pin numbers >= %d can be used\n", + config->dai_index, INTEL_ALH_DAI_INDEX_BASE); + return -EINVAL; + } + config->dai_index = data->dai_index - INTEL_ALH_DAI_INDEX_BASE; + } config->alh.stream_id = data->dai_data; } break; -- 2.50.1 From b682aa788e5f9f1ddacdfbb453e49fd3f4e83721 Mon Sep 17 00:00:00 2001 From: Ilya Zverev Date: Wed, 27 Nov 2024 15:44:20 +0200 Subject: [PATCH 11/16] ASoC: amd: yc: Add a quirk for microfone on Lenovo ThinkPad P14s Gen 5 21MES00B00 New ThinkPads need new quirk entries. Ilya has tested this one. Laptop product id is 21MES00B00, though the shorthand 21ME works. Closes: https://bugzilla.kernel.org/show_bug.cgi?id=219533 Cc: stable@vger.kernel.org Signed-off-by: Ilya Zverev Link: https://patch.msgid.link/20241127134420.14471-1-ilya@zverev.info Signed-off-by: Mark Brown --- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index facd82f0f251..e38c5885dadf 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -248,6 +248,13 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21M5"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21ME"), + } + }, { .driver_data = &acp6x_card, .matches = { -- 2.50.1 From ca0f79f0286046f6a91c099dc941cf7afae198d6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Nov 2024 08:26:45 +0100 Subject: [PATCH 12/16] ALSA: hda/realtek: Apply quirk for Medion E15433 Medion E15433 laptop wich ALC269VC (SSID 2782:1705) needs the same workaround for the missing speaker as another model. Link: https://bugzilla.suse.com/show_bug.cgi?id=1233298 Cc: Link: https://patch.msgid.link/20241128072646.15659-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- 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 d950666f9c74..4355282f5b2d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10952,6 +10952,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x2782, 0x0228, "Infinix ZERO BOOK 13", ALC269VB_FIXUP_INFINIX_ZERO_BOOK_13), SND_PCI_QUIRK(0x2782, 0x0232, "CHUWI CoreBook XPro", ALC269VB_FIXUP_CHUWI_COREBOOK_XPRO), SND_PCI_QUIRK(0x2782, 0x1701, "Infinix Y4 Max", ALC269VC_FIXUP_INFINIX_Y4_MAX), + SND_PCI_QUIRK(0x2782, 0x1705, "MEDION E15433", ALC269VC_FIXUP_INFINIX_Y4_MAX), SND_PCI_QUIRK(0x2782, 0x1707, "Vaio VJFE-ADL", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x2782, 0x4900, "MEDION E15443", ALC233_FIXUP_MEDION_MTL_SPK), SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), -- 2.50.1 From a7df7f909cec96e2fb7813a9b0b7e06a976983ab Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 28 Nov 2024 12:21:45 +0100 Subject: [PATCH 13/16] ALSA: hda: improve bass speaker support for ASUS Zenbook UM5606WA This hardware has ALC294 codec with speaker NID 0x17 and bass speaker NID 0x15. This patch removes DAC NID 0x06 (without volume control) from the connection list for bass speaker NID 0x15. Both speaker PINs are routed to DAC NID 0x03 with this change. Link: https://github.com/alsa-project/alsa-ucm-conf/issues/467 Signed-off-by: Jaroslav Kysela Link: https://patch.msgid.link/20241128112145.3409492-1-perex@perex.cz Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4355282f5b2d..2bf5c512ebaf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6491,6 +6491,16 @@ static void alc285_fixup_speaker2_to_dac1(struct hda_codec *codec, } } +/* disable DAC3 (0x06) selection on NID 0x15 - share Speaker/Bass Speaker DAC 0x03 */ +static void alc294_fixup_bass_speaker_15(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + static const hda_nid_t conn[] = { 0x02, 0x03 }; + snd_hda_override_conn_list(codec, 0x15, ARRAY_SIZE(conn), conn); + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -7773,6 +7783,7 @@ enum { ALC245_FIXUP_CLEVO_NOISY_MIC, ALC269_FIXUP_VAIO_VJFH52_MIC_NO_PRESENCE, ALC233_FIXUP_MEDION_MTL_SPK, + ALC294_FIXUP_BASS_SPEAKER_15, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -10081,6 +10092,10 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC294_FIXUP_BASS_SPEAKER_15] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc294_fixup_bass_speaker_15, + }, }; static const struct hda_quirk alc269_fixup_tbl[] = { @@ -10590,6 +10605,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606WA", ALC294_FIXUP_BASS_SPEAKER_15), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), -- 2.50.1 From aaa55faa2495320e44bc643a917c701f2cc89ee7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 28 Nov 2024 18:04:22 +0100 Subject: [PATCH 14/16] ALSA: seq: ump: Fix seq port updates per FB info notify update_port_infos() is called when a UMP FB Info update notification is received, and this function is supposed to update the attributes of the corresponding sequencer port. However, the function had a few issues and it brought to the incorrect states. Namely: - It tried to get a wrong sequencer info for the update without correcting the port number with the group-offset 1 - The loop exited immediately when a sequencer port isn't present; this ended up with the truncation if a sequencer port in the middle goes away This patch addresses those bugs. Fixes: 4a16a3af0571 ("ALSA: seq: ump: Handle FB info update") Link: https://patch.msgid.link/20241128170423.23351-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ump_client.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index e5d3f4d206bf..e956f17f3792 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -257,12 +257,12 @@ static void update_port_infos(struct seq_ump_client *client) continue; old->addr.client = client->seq_client; - old->addr.port = i; + old->addr.port = ump_group_to_seq_port(i); err = snd_seq_kernel_client_ctl(client->seq_client, SNDRV_SEQ_IOCTL_GET_PORT_INFO, old); if (err < 0) - return; + continue; fill_port_info(new, client, &client->ump->groups[i]); if (old->capability == new->capability && !strcmp(old->name, new->name)) @@ -271,7 +271,7 @@ static void update_port_infos(struct seq_ump_client *client) SNDRV_SEQ_IOCTL_SET_PORT_INFO, new); if (err < 0) - return; + continue; /* notify to system port */ snd_seq_system_client_ev_port_change(client->seq_client, i); } -- 2.50.1 From 3978d53df7236f0a517c2abeb43ddf6ac162cdd8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Nov 2024 10:45:42 +0100 Subject: [PATCH 15/16] ALSA: ump: Don't open legacy substream for an inactive group When a UMP Group is inactive, we shouldn't allow users to access it via the legacy MIDI access. Add the group active flag check and return -ENODEV if it's inactive. Link: https://patch.msgid.link/20241129094546.32119-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/ump.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/ump.c b/sound/core/ump.c index 6d0aac6c763f..9f9ad106107f 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -1087,6 +1087,8 @@ static int snd_ump_legacy_open(struct snd_rawmidi_substream *substream) guard(mutex)(&ump->open_mutex); if (ump->legacy_substreams[dir][group]) return -EBUSY; + if (!ump->groups[group].active) + return -ENODEV; if (dir == SNDRV_RAWMIDI_STREAM_OUTPUT) { if (!ump->legacy_out_opens) { err = snd_rawmidi_kernel_open(&ump->core, 0, -- 2.50.1 From e29e504e7890b9ee438ca6370d0180d607c473f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 29 Nov 2024 10:45:43 +0100 Subject: [PATCH 16/16] ALSA: ump: Indicate the inactive group in legacy substream names Since the legacy rawmidi has no proper way to know the inactive group, indicate it in the rawmidi substream names with "[Inactive]" suffix when the corresponding UMP group is inactive. Link: https://patch.msgid.link/20241129094546.32119-3-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/ump.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/core/ump.c b/sound/core/ump.c index 9f9ad106107f..7b00a957e548 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -1256,8 +1256,9 @@ static void fill_substream_names(struct snd_ump_endpoint *ump, name = ump->groups[idx].name; if (!*name) name = ump->info.name; - snprintf(s->name, sizeof(s->name), "Group %d (%.16s)", - idx + 1, name); + snprintf(s->name, sizeof(s->name), "Group %d (%.16s)%s", + idx + 1, name, + ump->groups[idx].active ? "" : " [Inactive]"); } } -- 2.50.1