From 495000a38634e640e2fd02f7e4f1512ccc92d770 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 24 May 2024 17:11:46 +0200 Subject: [PATCH 01/26] ALSA: core: Remove debugfs at disconnection The card-specific debugfs entries are removed at the last stage of card free phase, and it's performed after synchronization of the closes of all opened fds. This works fine for most cases, but it can be potentially problematic for a hotplug device like USB-audio. Due to the nature of snd_card_free_when_closed(), the card free isn't called immediately after the driver removal for a hotplug device, but it's left until the last fd is closed. It implies that the card debugfs entries also remain. Meanwhile, when a new device is inserted before the last close and the very same card slot is assigned, the driver tries to create the card debugfs root again on the very same path. This conflicts with the remaining entry, and results in the kernel warning such as: debugfs: Directory 'card0' with parent 'sound' already present! with the missing debugfs entry afterwards. For avoiding such conflicts, remove debugfs entries at the device disconnection phase instead. The jack kctl debugfs entries get removed in snd_jack_dev_disconnect() instead of each kctl private_free. Fixes: 2d670ea2bd53 ("ALSA: jack: implement software jack injection via debugfs") Link: https://lore.kernel.org/r/20240524151256.32521-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/init.c | 9 +++++---- sound/core/jack.c | 21 ++++++++++++++------- 2 files changed, 19 insertions(+), 11 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 4e52bbe32786..b9b708cf980d 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -537,6 +537,11 @@ void snd_card_disconnect(struct snd_card *card) synchronize_irq(card->sync_irq); snd_info_card_disconnect(card); +#ifdef CONFIG_SND_DEBUG + debugfs_remove(card->debugfs_root); + card->debugfs_root = NULL; +#endif + if (card->registered) { device_del(&card->card_dev); card->registered = false; @@ -586,10 +591,6 @@ static int snd_card_do_free(struct snd_card *card) dev_warn(card->dev, "unable to free card info\n"); /* Not fatal error */ } -#ifdef CONFIG_SND_DEBUG - debugfs_remove(card->debugfs_root); - card->debugfs_root = NULL; -#endif if (card->release_completion) complete(card->release_completion); if (!card->managed) diff --git a/sound/core/jack.c b/sound/core/jack.c index e08b2c4fbd1a..e4bcecdf89b7 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -37,11 +37,15 @@ static const int jack_switch_types[SND_JACK_SWITCH_TYPES] = { }; #endif /* CONFIG_SND_JACK_INPUT_DEV */ +static void snd_jack_remove_debugfs(struct snd_jack *jack); + static int snd_jack_dev_disconnect(struct snd_device *device) { -#ifdef CONFIG_SND_JACK_INPUT_DEV struct snd_jack *jack = device->device_data; + snd_jack_remove_debugfs(jack); + +#ifdef CONFIG_SND_JACK_INPUT_DEV guard(mutex)(&jack->input_dev_lock); if (!jack->input_dev) return 0; @@ -381,10 +385,14 @@ static int snd_jack_debugfs_add_inject_node(struct snd_jack *jack, return 0; } -static void snd_jack_debugfs_clear_inject_node(struct snd_jack_kctl *jack_kctl) +static void snd_jack_remove_debugfs(struct snd_jack *jack) { - debugfs_remove(jack_kctl->jack_debugfs_root); - jack_kctl->jack_debugfs_root = NULL; + struct snd_jack_kctl *jack_kctl; + + list_for_each_entry(jack_kctl, &jack->kctl_list, list) { + debugfs_remove(jack_kctl->jack_debugfs_root); + jack_kctl->jack_debugfs_root = NULL; + } } #else /* CONFIG_SND_JACK_INJECTION_DEBUG */ static int snd_jack_debugfs_add_inject_node(struct snd_jack *jack, @@ -393,7 +401,7 @@ static int snd_jack_debugfs_add_inject_node(struct snd_jack *jack, return 0; } -static void snd_jack_debugfs_clear_inject_node(struct snd_jack_kctl *jack_kctl) +static void snd_jack_remove_debugfs(struct snd_jack *jack) { } #endif /* CONFIG_SND_JACK_INJECTION_DEBUG */ @@ -404,7 +412,6 @@ static void snd_jack_kctl_private_free(struct snd_kcontrol *kctl) jack_kctl = kctl->private_data; if (jack_kctl) { - snd_jack_debugfs_clear_inject_node(jack_kctl); list_del(&jack_kctl->list); kfree(jack_kctl); } @@ -497,8 +504,8 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, .dev_free = snd_jack_dev_free, #ifdef CONFIG_SND_JACK_INPUT_DEV .dev_register = snd_jack_dev_register, - .dev_disconnect = snd_jack_dev_disconnect, #endif /* CONFIG_SND_JACK_INPUT_DEV */ + .dev_disconnect = snd_jack_dev_disconnect, }; if (initial_kctl) { From 2be46155d792d629e8fe3188c2cde176833afe36 Mon Sep 17 00:00:00 2001 From: "Luke D. Jones" Date: Sun, 26 May 2024 21:10:32 +1200 Subject: [PATCH 02/26] ALSA: hda/realtek: Adjust G814JZR to use SPI init for amp The 2024 ASUS ROG G814J model is much the same as the 2023 model and the 2023 16" version. We can use the same Cirrus Amp quirk. Fixes: 811dd426a9b1 ("ALSA: hda/realtek: Add quirks for Asus ROG 2024 laptops using CS35L41") Signed-off-by: Luke D. Jones Link: https://lore.kernel.org/r/20240526091032.114545-1-luke@ljones.dev Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e3c0b9d5552d..aa76d1c88589 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10310,7 +10310,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x3a20, "ASUS G614JZR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a30, "ASUS G814JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x3a40, "ASUS G814JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x3a50, "ASUS G834JYR/JZR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x3a60, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), From 797c525e85d1e44cf0e6f338890e8e0c661f524a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 27 May 2024 11:08:40 +0100 Subject: [PATCH 03/26] ASoC: cs42l43: Only restrict 44.1kHz for the ASP The SoundWire interface can always support 44.1kHz using flow controlled mode, and whether the ASP is in master mode should obviously only affect the ASP. Update cs42l43_startup() to only restrict the rates for the ASP DAI. Fixes: fc918cbe874e ("ASoC: cs42l43: Add support for the cs42l43") Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240527100840.439832-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l43.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 94685449f0f4..92674314227c 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -310,8 +310,9 @@ static int cs42l43_startup(struct snd_pcm_substream *substream, struct snd_soc_d struct snd_soc_component *component = dai->component; struct cs42l43_codec *priv = snd_soc_component_get_drvdata(component); struct cs42l43 *cs42l43 = priv->core; - int provider = !!regmap_test_bits(cs42l43->regmap, CS42L43_ASP_CLK_CONFIG2, - CS42L43_ASP_MASTER_MODE_MASK); + int provider = !dai->id || !!regmap_test_bits(cs42l43->regmap, + CS42L43_ASP_CLK_CONFIG2, + CS42L43_ASP_MASTER_MODE_MASK); if (provider) priv->constraint.mask = CS42L43_PROVIDER_RATE_MASK; From 8d34c12e8751fe180157fbb34a758ed4eede3806 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 27 May 2024 11:02:37 +0100 Subject: [PATCH 04/26] ASoC: wm_adsp: Add missing MODULE_DESCRIPTION() wm_adsp is built as a separate module and as such should include a MODULE_DESCRIPTION() macro. Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240527100237.430240-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index c9d9a7b28efb..68d2d6444533 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2085,5 +2085,6 @@ static const struct cs_dsp_client_ops wm_adsp2_client_ops = { .watchdog_expired = wm_adsp_fatal_error, }; +MODULE_DESCRIPTION("Cirrus Logic ASoC DSP Support"); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(FW_CS_DSP); From d5d2a5dacbc8ea4386071ce243c43ea0dac23cb8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 27 May 2024 11:13:26 +0100 Subject: [PATCH 05/26] MAINTAINERS: Remove James Schulman from Cirrus audio maintainers James no longer works for Cirrus Logic, remove him from the list of maintainers for the Cirrus audio CODEC drivers. Signed-off-by: Charles Keepax Link: https://msgid.link/r/20240527101326.440345-1-ckeepax@opensource.cirrus.com Signed-off-by: Mark Brown --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index d6c90161c7bf..a6a011792167 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -5187,7 +5187,6 @@ F: Documentation/devicetree/bindings/media/i2c/chrontel,ch7322.yaml F: drivers/media/cec/i2c/ch7322.c CIRRUS LOGIC AUDIO CODEC DRIVERS -M: James Schulman M: David Rhodes M: Richard Fitzgerald L: alsa-devel@alsa-project.org (moderated for non-subscribers) From e30a942861b540e056425a8e31ba801de1ed4f25 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 27 May 2024 14:44:11 -0500 Subject: [PATCH 06/26] ASoC: SOF: stream-ipc: remove unnecessary MODULE_LICENSE MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit This file is part of the snd-sof module, there's no reason to re-add the MODULE_LICENSE here. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Daniel Baluta Reviewed-by: Péter Ujfalusi Link: https://msgid.link/r/20240527194414.166156-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/stream-ipc.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sof/stream-ipc.c b/sound/soc/sof/stream-ipc.c index eb71303aa24c..794c7bbccbaf 100644 --- a/sound/soc/sof/stream-ipc.c +++ b/sound/soc/sof/stream-ipc.c @@ -125,5 +125,3 @@ int sof_stream_pcm_close(struct snd_sof_dev *sdev, return 0; } EXPORT_SYMBOL(sof_stream_pcm_close); - -MODULE_LICENSE("Dual BSD/GPL"); From b88056df4fcb7b5930d6ee3fef494e8729dcf2b2 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 27 May 2024 14:44:12 -0500 Subject: [PATCH 07/26] ASoC: SOF: AMD: group all module related information MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The module information is spread across files, group in a single location. For maintenability and alignment, the arbitrary Intel convention is used with the following order: MODULE_LICENSE MODULE_DESCRIPTION MODULE_IMPORT Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Daniel Baluta Reviewed-by: Péter Ujfalusi Link: https://msgid.link/r/20240527194414.166156-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp63.c | 4 ---- sound/soc/sof/amd/pci-acp63.c | 1 + sound/soc/sof/amd/pci-rmb.c | 1 + sound/soc/sof/amd/pci-rn.c | 1 + sound/soc/sof/amd/pci-vangogh.c | 1 + sound/soc/sof/amd/rembrandt.c | 4 ---- sound/soc/sof/amd/renoir.c | 4 ---- sound/soc/sof/amd/vangogh.c | 4 ---- 8 files changed, 4 insertions(+), 16 deletions(-) diff --git a/sound/soc/sof/amd/acp63.c b/sound/soc/sof/amd/acp63.c index 9fb645079c3a..9e6eb4bfc805 100644 --- a/sound/soc/sof/amd/acp63.c +++ b/sound/soc/sof/amd/acp63.c @@ -140,7 +140,3 @@ int sof_acp63_ops_init(struct snd_sof_dev *sdev) return 0; } - -MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); -MODULE_DESCRIPTION("ACP63 SOF Driver"); -MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index eeaa12cceb23..fc8984447365 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -109,5 +109,6 @@ static struct pci_driver snd_sof_pci_amd_acp63_driver = { module_pci_driver(snd_sof_pci_amd_acp63_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("ACP63 SOF Driver"); MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index 2f288545c426..4bc30951f8b0 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -99,5 +99,6 @@ static struct pci_driver snd_sof_pci_amd_rmb_driver = { module_pci_driver(snd_sof_pci_amd_rmb_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("REMBRANDT SOF Driver"); MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index a0195e9b400c..e08875bdfa8b 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -103,5 +103,6 @@ static struct pci_driver snd_sof_pci_amd_rn_driver = { module_pci_driver(snd_sof_pci_amd_rn_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("RENOIR SOF Driver"); MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c index 5cd3ac84752f..16eb2994fbab 100644 --- a/sound/soc/sof/amd/pci-vangogh.c +++ b/sound/soc/sof/amd/pci-vangogh.c @@ -101,5 +101,6 @@ static struct pci_driver snd_sof_pci_amd_vgh_driver = { module_pci_driver(snd_sof_pci_amd_vgh_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("VANGOGH SOF Driver"); MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/amd/rembrandt.c b/sound/soc/sof/amd/rembrandt.c index f1d1ba57ab3a..076f2f05a95c 100644 --- a/sound/soc/sof/amd/rembrandt.c +++ b/sound/soc/sof/amd/rembrandt.c @@ -140,7 +140,3 @@ int sof_rembrandt_ops_init(struct snd_sof_dev *sdev) return 0; } - -MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); -MODULE_DESCRIPTION("REMBRANDT SOF Driver"); -MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/amd/renoir.c b/sound/soc/sof/amd/renoir.c index 47b863f6258c..aa2d24dac6f5 100644 --- a/sound/soc/sof/amd/renoir.c +++ b/sound/soc/sof/amd/renoir.c @@ -115,7 +115,3 @@ int sof_renoir_ops_init(struct snd_sof_dev *sdev) return 0; } - -MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); -MODULE_DESCRIPTION("RENOIR SOF Driver"); -MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/amd/vangogh.c b/sound/soc/sof/amd/vangogh.c index bc6ffdb5471a..61372958c09d 100644 --- a/sound/soc/sof/amd/vangogh.c +++ b/sound/soc/sof/amd/vangogh.c @@ -161,7 +161,3 @@ int sof_vangogh_ops_init(struct snd_sof_dev *sdev) return 0; } - -MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); -MODULE_DESCRIPTION("VANGOGH SOF Driver"); -MODULE_LICENSE("Dual BSD/GPL"); From 06a2315da0b02db4f2115bc9253daa270571e389 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 27 May 2024 14:44:13 -0500 Subject: [PATCH 08/26] ASoC: SOF: reorder MODULE_ definitions MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Follow the arbitrary Intel convention order to allow for easier grep. MODULE_LICENSE MODULE_DESCRIPTION MODULE_IMPORT Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Daniel Baluta Reviewed-by: Péter Ujfalusi Link: https://msgid.link/r/20240527194414.166156-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/amd/acp-common.c | 4 ++-- sound/soc/sof/amd/acp.c | 2 +- sound/soc/sof/core.c | 2 +- sound/soc/sof/nocodec.c | 2 +- sound/soc/sof/sof-client-ipc-flood-test.c | 2 +- sound/soc/sof/sof-client-ipc-kernel-injector.c | 2 +- sound/soc/sof/sof-client-ipc-msg-injector.c | 2 +- sound/soc/sof/sof-client-probes.c | 2 +- sound/soc/sof/xtensa/core.c | 2 +- 9 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/sof/amd/acp-common.c b/sound/soc/sof/amd/acp-common.c index b26fa471b431..81bb93e98358 100644 --- a/sound/soc/sof/amd/acp-common.c +++ b/sound/soc/sof/amd/acp-common.c @@ -258,8 +258,8 @@ const struct snd_sof_dsp_ops sof_acp_common_ops = { }; EXPORT_SYMBOL_NS(sof_acp_common_ops, SND_SOC_SOF_AMD_COMMON); +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("ACP SOF COMMON Driver"); MODULE_IMPORT_NS(SND_SOC_SOF_AMD_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_IMPORT_NS(SOUNDWIRE_AMD_INIT); -MODULE_DESCRIPTION("ACP SOF COMMON Driver"); -MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/amd/acp.c b/sound/soc/sof/amd/acp.c index c12c7f820529..74fd5f2b148b 100644 --- a/sound/soc/sof/amd/acp.c +++ b/sound/soc/sof/amd/acp.c @@ -801,7 +801,7 @@ void amd_sof_acp_remove(struct snd_sof_dev *sdev) } EXPORT_SYMBOL_NS(amd_sof_acp_remove, SND_SOC_SOF_AMD_COMMON); +MODULE_LICENSE("Dual BSD/GPL"); MODULE_DESCRIPTION("AMD ACP sof driver"); MODULE_IMPORT_NS(SOUNDWIRE_AMD_INIT); MODULE_IMPORT_NS(SND_AMD_SOUNDWIRE_ACPI); -MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 0a4917136ff9..83fe0401baf8 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -769,7 +769,7 @@ void sof_machine_unregister(struct snd_sof_dev *sdev, void *pdata) EXPORT_SYMBOL(sof_machine_unregister); MODULE_AUTHOR("Liam Girdwood"); -MODULE_DESCRIPTION("Sound Open Firmware (SOF) Core"); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("Sound Open Firmware (SOF) Core"); MODULE_ALIAS("platform:sof-audio"); MODULE_IMPORT_NS(SND_SOC_SOF_CLIENT); diff --git a/sound/soc/sof/nocodec.c b/sound/soc/sof/nocodec.c index fdcbe33d3dcf..b12b3d865ae3 100644 --- a/sound/soc/sof/nocodec.c +++ b/sound/soc/sof/nocodec.c @@ -110,7 +110,7 @@ static struct platform_driver sof_nocodec_audio = { }; module_platform_driver(sof_nocodec_audio) +MODULE_LICENSE("Dual BSD/GPL"); MODULE_DESCRIPTION("ASoC sof nocodec"); MODULE_AUTHOR("Liam Girdwood"); -MODULE_LICENSE("Dual BSD/GPL"); MODULE_ALIAS("platform:sof-nocodec"); diff --git a/sound/soc/sof/sof-client-ipc-flood-test.c b/sound/soc/sof/sof-client-ipc-flood-test.c index 435614926092..e7d2001140e8 100644 --- a/sound/soc/sof/sof-client-ipc-flood-test.c +++ b/sound/soc/sof/sof-client-ipc-flood-test.c @@ -394,6 +394,6 @@ static struct auxiliary_driver sof_ipc_flood_client_drv = { module_auxiliary_driver(sof_ipc_flood_client_drv); -MODULE_DESCRIPTION("SOF IPC Flood Test Client Driver"); MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SOF IPC Flood Test Client Driver"); MODULE_IMPORT_NS(SND_SOC_SOF_CLIENT); diff --git a/sound/soc/sof/sof-client-ipc-kernel-injector.c b/sound/soc/sof/sof-client-ipc-kernel-injector.c index 6973b6690df4..d3f541069b24 100644 --- a/sound/soc/sof/sof-client-ipc-kernel-injector.c +++ b/sound/soc/sof/sof-client-ipc-kernel-injector.c @@ -157,6 +157,6 @@ static struct auxiliary_driver sof_msg_inject_client_drv = { module_auxiliary_driver(sof_msg_inject_client_drv); -MODULE_DESCRIPTION("SOF IPC Kernel Injector Client Driver"); MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SOF IPC Kernel Injector Client Driver"); MODULE_IMPORT_NS(SND_SOC_SOF_CLIENT); diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index af22e6421029..d0f8beb9d000 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -335,6 +335,6 @@ static struct auxiliary_driver sof_msg_inject_client_drv = { module_auxiliary_driver(sof_msg_inject_client_drv); -MODULE_DESCRIPTION("SOF IPC Message Injector Client Driver"); MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SOF IPC Message Injector Client Driver"); MODULE_IMPORT_NS(SND_SOC_SOF_CLIENT); diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index b8f297307565..ccc7d38ddc38 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -540,6 +540,6 @@ static struct auxiliary_driver sof_probes_client_drv = { module_auxiliary_driver(sof_probes_client_drv); -MODULE_DESCRIPTION("SOF Probes Client Driver"); MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("SOF Probes Client Driver"); MODULE_IMPORT_NS(SND_SOC_SOF_CLIENT); diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c index ccbc3fcdadd5..3cf8c84beff9 100644 --- a/sound/soc/sof/xtensa/core.c +++ b/sound/soc/sof/xtensa/core.c @@ -151,5 +151,5 @@ const struct dsp_arch_ops sof_xtensa_arch_ops = { }; EXPORT_SYMBOL_NS(sof_xtensa_arch_ops, SND_SOC_SOF_XTENSA); -MODULE_DESCRIPTION("SOF Xtensa DSP support"); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF Xtensa DSP support"); From 3ff78451b8e446e9a548b98a0d4dd8d24dc5780b Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 27 May 2024 14:44:14 -0500 Subject: [PATCH 09/26] ASoC: SOF: add missing MODULE_DESCRIPTION() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit MODULE_DESCRIPTION() was optional until it became mandatory and flagged as an error by 'make W=1'. Reported-by: Andy Shevchenko Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Daniel Baluta Reviewed-by: Péter Ujfalusi Link: https://msgid.link/r/20240527194414.166156-5-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/imx/imx-common.c | 1 + sound/soc/sof/imx/imx8.c | 3 ++- sound/soc/sof/imx/imx8m.c | 3 ++- sound/soc/sof/imx/imx8ulp.c | 3 ++- sound/soc/sof/intel/atom.c | 1 + sound/soc/sof/intel/bdw.c | 1 + sound/soc/sof/intel/byt.c | 1 + sound/soc/sof/intel/hda-codec.c | 1 + sound/soc/sof/intel/hda-ctrl.c | 1 + sound/soc/sof/intel/hda-mlink.c | 1 + sound/soc/sof/intel/hda.c | 1 + sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-lnl.c | 1 + sound/soc/sof/intel/pci-mtl.c | 1 + sound/soc/sof/intel/pci-skl.c | 1 + sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/sof/intel/pci-tng.c | 1 + sound/soc/sof/mediatek/mt8186/mt8186.c | 3 ++- sound/soc/sof/mediatek/mt8195/mt8195.c | 3 ++- sound/soc/sof/mediatek/mtk-adsp-common.c | 1 + sound/soc/sof/sof-acpi-dev.c | 1 + sound/soc/sof/sof-of-dev.c | 1 + sound/soc/sof/sof-pci-dev.c | 1 + sound/soc/sof/sof-utils.c | 1 + 26 files changed, 31 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/imx/imx-common.c b/sound/soc/sof/imx/imx-common.c index 2981aea123d9..fce6d9cf6a6b 100644 --- a/sound/soc/sof/imx/imx-common.c +++ b/sound/soc/sof/imx/imx-common.c @@ -75,3 +75,4 @@ void imx8_dump(struct snd_sof_dev *sdev, u32 flags) EXPORT_SYMBOL(imx8_dump); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF helpers for IMX platforms"); diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 3021dc87ab5a..9f24e3c283dd 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -667,5 +667,6 @@ static struct platform_driver snd_sof_of_imx8_driver = { }; module_platform_driver(snd_sof_of_imx8_driver); -MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for IMX8 platforms"); +MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 4ed415f04345..1c7019c3cbd3 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -514,5 +514,6 @@ static struct platform_driver snd_sof_of_imx8m_driver = { }; module_platform_driver(snd_sof_of_imx8m_driver); -MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for IMX8M platforms"); +MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 8adfdd00413a..2585b1beef23 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -516,5 +516,6 @@ static struct platform_driver snd_sof_of_imx8ulp_driver = { }; module_platform_driver(snd_sof_of_imx8ulp_driver); -MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for IMX8ULP platforms"); +MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); diff --git a/sound/soc/sof/intel/atom.c b/sound/soc/sof/intel/atom.c index 86af4e9a716e..3505ac3a1b14 100644 --- a/sound/soc/sof/intel/atom.c +++ b/sound/soc/sof/intel/atom.c @@ -418,3 +418,4 @@ void atom_set_mach_params(struct snd_soc_acpi_mach *mach, EXPORT_SYMBOL_NS(atom_set_mach_params, SND_SOC_SOF_INTEL_ATOM_HIFI_EP); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for Atom platforms"); diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 3262286a9a9d..7f18080e4e19 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -694,6 +694,7 @@ static struct platform_driver snd_sof_acpi_intel_bdw_driver = { module_platform_driver(snd_sof_acpi_intel_bdw_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for Broadwell platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HIFI_EP_IPC); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_IMPORT_NS(SND_SOC_SOF_ACPI_DEV); diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index d78d11d4cfbf..7a57e162fb1c 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -475,6 +475,7 @@ static struct platform_driver snd_sof_acpi_intel_byt_driver = { module_platform_driver(snd_sof_acpi_intel_byt_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for Baytrail/Cherrytrail"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HIFI_EP_IPC); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_IMPORT_NS(SND_SOC_SOF_ACPI_DEV); diff --git a/sound/soc/sof/intel/hda-codec.c b/sound/soc/sof/intel/hda-codec.c index da3db3ed379e..dc46888faa0d 100644 --- a/sound/soc/sof/intel/hda-codec.c +++ b/sound/soc/sof/intel/hda-codec.c @@ -457,3 +457,4 @@ EXPORT_SYMBOL_NS_GPL(hda_codec_i915_exit, SND_SOC_SOF_HDA_AUDIO_CODEC_I915); #endif MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for HDaudio codecs"); diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 262b482dc0a8..b9a02750ce61 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -328,6 +328,7 @@ void hda_dsp_ctrl_stop_chip(struct snd_sof_dev *sdev) } MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF helpers for HDaudio platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_MLINK); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC_I915); diff --git a/sound/soc/sof/intel/hda-mlink.c b/sound/soc/sof/intel/hda-mlink.c index 04bbc5c9904c..9a3559c78b62 100644 --- a/sound/soc/sof/intel/hda-mlink.c +++ b/sound/soc/sof/intel/hda-mlink.c @@ -972,3 +972,4 @@ EXPORT_SYMBOL_NS(hdac_bus_eml_enable_offload, SND_SOC_SOF_HDA_MLINK); #endif MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for HDaudio multi-link"); diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index e6a38de0a0aa..dead1c19558b 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1522,6 +1522,7 @@ void hda_unregister_clients(struct snd_sof_dev *sdev) } MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for HDaudio platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC); MODULE_IMPORT_NS(SND_SOC_SOF_HDA_AUDIO_CODEC_I915); diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c index df6d897da290..f006dcf5458a 100644 --- a/sound/soc/sof/intel/pci-apl.c +++ b/sound/soc/sof/intel/pci-apl.c @@ -105,6 +105,7 @@ static struct pci_driver snd_sof_pci_intel_apl_driver = { module_pci_driver(snd_sof_pci_intel_apl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for ApolloLake platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c index a39fa3657d55..a8406342f08b 100644 --- a/sound/soc/sof/intel/pci-cnl.c +++ b/sound/soc/sof/intel/pci-cnl.c @@ -143,6 +143,7 @@ static struct pci_driver snd_sof_pci_intel_cnl_driver = { module_pci_driver(snd_sof_pci_intel_cnl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for CannonLake platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c index 9f1fe47475fb..25effca50d9f 100644 --- a/sound/soc/sof/intel/pci-icl.c +++ b/sound/soc/sof/intel/pci-icl.c @@ -108,6 +108,7 @@ static struct pci_driver snd_sof_pci_intel_icl_driver = { module_pci_driver(snd_sof_pci_intel_icl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for IceLake platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL); diff --git a/sound/soc/sof/intel/pci-lnl.c b/sound/soc/sof/intel/pci-lnl.c index 68e5c90151b2..602c574064eb 100644 --- a/sound/soc/sof/intel/pci-lnl.c +++ b/sound/soc/sof/intel/pci-lnl.c @@ -70,6 +70,7 @@ static struct pci_driver snd_sof_pci_intel_lnl_driver = { module_pci_driver(snd_sof_pci_intel_lnl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for LunarLake platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_MTL); diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c index c685cb8d6171..8cb0333c033e 100644 --- a/sound/soc/sof/intel/pci-mtl.c +++ b/sound/soc/sof/intel/pci-mtl.c @@ -133,6 +133,7 @@ static struct pci_driver snd_sof_pci_intel_mtl_driver = { module_pci_driver(snd_sof_pci_intel_mtl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for MeteorLake platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c index 862da8009543..8ca0231d7e4f 100644 --- a/sound/soc/sof/intel/pci-skl.c +++ b/sound/soc/sof/intel/pci-skl.c @@ -89,6 +89,7 @@ static struct pci_driver snd_sof_pci_intel_skl_driver = { module_pci_driver(snd_sof_pci_intel_skl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for SkyLake platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index f73bb47cd79e..ebe1a7d16689 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -317,6 +317,7 @@ static struct pci_driver snd_sof_pci_intel_tgl_driver = { module_pci_driver(snd_sof_pci_intel_tgl_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for TigerLake platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_GENERIC); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HDA_COMMON); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_CNL); diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index 5c3069588bb7..1375c393827e 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -244,6 +244,7 @@ static struct pci_driver snd_sof_pci_intel_tng_driver = { module_pci_driver(snd_sof_pci_intel_tng_driver); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for Tangier platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_INTEL_HIFI_EP_IPC); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_IMPORT_NS(SND_SOC_SOF_PCI_DEV); diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index c63e0d2f4b96..bea1b9d9ca28 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -666,6 +666,7 @@ static struct platform_driver snd_sof_of_mt8186_driver = { }; module_platform_driver(snd_sof_of_mt8186_driver); +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for MT8186/MT8188 platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_IMPORT_NS(SND_SOC_SOF_MTK_COMMON); -MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index fc1c016104ae..31dc98d1b1d8 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -619,6 +619,7 @@ static struct platform_driver snd_sof_of_mt8195_driver = { }; module_platform_driver(snd_sof_of_mt8195_driver); +MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for MTL 8195 platforms"); MODULE_IMPORT_NS(SND_SOC_SOF_XTENSA); MODULE_IMPORT_NS(SND_SOC_SOF_MTK_COMMON); -MODULE_LICENSE("Dual BSD/GPL"); diff --git a/sound/soc/sof/mediatek/mtk-adsp-common.c b/sound/soc/sof/mediatek/mtk-adsp-common.c index de8dbe27cd0d..20bcf5590eb8 100644 --- a/sound/soc/sof/mediatek/mtk-adsp-common.c +++ b/sound/soc/sof/mediatek/mtk-adsp-common.c @@ -82,3 +82,4 @@ void mtk_adsp_dump(struct snd_sof_dev *sdev, u32 flags) EXPORT_SYMBOL(mtk_adsp_dump); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF helpers for MTK ADSP platforms"); diff --git a/sound/soc/sof/sof-acpi-dev.c b/sound/soc/sof/sof-acpi-dev.c index 2d96d00f1c44..b196b2b74c26 100644 --- a/sound/soc/sof/sof-acpi-dev.c +++ b/sound/soc/sof/sof-acpi-dev.c @@ -100,3 +100,4 @@ void sof_acpi_remove(struct platform_device *pdev) EXPORT_SYMBOL_NS(sof_acpi_remove, SND_SOC_SOF_ACPI_DEV); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for ACPI platforms"); diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index b9a499e92b9a..71f7153cf79c 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -93,3 +93,4 @@ void sof_of_shutdown(struct platform_device *pdev) EXPORT_SYMBOL(sof_of_shutdown); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for OF/DT platforms"); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 4365405783e6..38f2187da5de 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -304,3 +304,4 @@ void sof_pci_shutdown(struct pci_dev *pci) EXPORT_SYMBOL_NS(sof_pci_shutdown, SND_SOC_SOF_PCI_DEV); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF support for PCI platforms"); diff --git a/sound/soc/sof/sof-utils.c b/sound/soc/sof/sof-utils.c index cad041bf56cc..44608682e9f8 100644 --- a/sound/soc/sof/sof-utils.c +++ b/sound/soc/sof/sof-utils.c @@ -73,3 +73,4 @@ int snd_sof_create_page_table(struct device *dev, EXPORT_SYMBOL(snd_sof_create_page_table); MODULE_LICENSE("Dual BSD/GPL"); +MODULE_DESCRIPTION("SOF utils"); From 8a42886cae307663f3f999846926bd6e64392000 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 May 2024 17:18:49 +0200 Subject: [PATCH 10/26] ALSA: seq: Fix missing bank setup between MIDI1/MIDI2 UMP conversion When a UMP packet is converted between MIDI1 and MIDI2 protocols, the bank selection may be lost. The conversion from MIDI1 to MIDI2 needs the encoding of the bank into UMP_MSG_STATUS_PROGRAM bits, while the conversion from MIDI2 to MIDI1 needs the extraction from that instead. This patch implements the missing bank selection mechanism in those conversions. Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events") Link: https://lore.kernel.org/r/20240527151852.29036-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ump_convert.c | 38 ++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index ee6ac649df83..c21be87f5da9 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -368,6 +368,7 @@ static int cvt_ump_midi1_to_midi2(struct snd_seq_client *dest, struct snd_seq_ump_event ev_cvt; const union snd_ump_midi1_msg *midi1 = (const union snd_ump_midi1_msg *)event->ump; union snd_ump_midi2_msg *midi2 = (union snd_ump_midi2_msg *)ev_cvt.ump; + struct snd_seq_ump_midi2_bank *cc; ev_cvt = *event; memset(&ev_cvt.ump, 0, sizeof(ev_cvt.ump)); @@ -387,11 +388,29 @@ static int cvt_ump_midi1_to_midi2(struct snd_seq_client *dest, midi2->paf.data = upscale_7_to_32bit(midi1->paf.data); break; case UMP_MSG_STATUS_CC: + cc = &dest_port->midi2_bank[midi1->note.channel]; + switch (midi1->cc.index) { + case UMP_CC_BANK_SELECT: + cc->bank_set = 1; + cc->cc_bank_msb = midi1->cc.data; + return 0; // skip + case UMP_CC_BANK_SELECT_LSB: + cc->bank_set = 1; + cc->cc_bank_lsb = midi1->cc.data; + return 0; // skip + } midi2->cc.index = midi1->cc.index; midi2->cc.data = upscale_7_to_32bit(midi1->cc.data); break; case UMP_MSG_STATUS_PROGRAM: midi2->pg.program = midi1->pg.program; + cc = &dest_port->midi2_bank[midi1->note.channel]; + if (cc->bank_set) { + midi2->pg.bank_valid = 1; + midi2->pg.bank_msb = cc->cc_bank_msb; + midi2->pg.bank_lsb = cc->cc_bank_lsb; + cc->bank_set = 0; + } break; case UMP_MSG_STATUS_CHANNEL_PRESSURE: midi2->caf.data = upscale_7_to_32bit(midi1->caf.data); @@ -419,6 +438,7 @@ static int cvt_ump_midi2_to_midi1(struct snd_seq_client *dest, struct snd_seq_ump_event ev_cvt; union snd_ump_midi1_msg *midi1 = (union snd_ump_midi1_msg *)ev_cvt.ump; const union snd_ump_midi2_msg *midi2 = (const union snd_ump_midi2_msg *)event->ump; + int err; u16 v; ev_cvt = *event; @@ -443,6 +463,24 @@ static int cvt_ump_midi2_to_midi1(struct snd_seq_client *dest, midi1->cc.data = downscale_32_to_7bit(midi2->cc.data); break; case UMP_MSG_STATUS_PROGRAM: + if (midi2->pg.bank_valid) { + midi1->cc.status = UMP_MSG_STATUS_CC; + midi1->cc.index = UMP_CC_BANK_SELECT; + midi1->cc.data = midi2->pg.bank_msb; + err = __snd_seq_deliver_single_event(dest, dest_port, + (struct snd_seq_event *)&ev_cvt, + atomic, hop); + if (err < 0) + return err; + midi1->cc.index = UMP_CC_BANK_SELECT_LSB; + midi1->cc.data = midi2->pg.bank_lsb; + err = __snd_seq_deliver_single_event(dest, dest_port, + (struct snd_seq_event *)&ev_cvt, + atomic, hop); + if (err < 0) + return err; + midi1->note.status = midi2->note.status; + } midi1->pg.program = midi2->pg.program; break; case UMP_MSG_STATUS_CHANNEL_PRESSURE: From a200df7deb3186cd7b55abb77ab96dfefb8a4f09 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 27 May 2024 17:18:50 +0200 Subject: [PATCH 11/26] ALSA: seq: Don't clear bank selection at event -> UMP MIDI2 conversion The current code to convert from a legacy sequencer event to UMP MIDI2 clears the bank selection at each time the program change is submitted. This is confusing and may lead to incorrect bank values tranmitted to the destination in the end. Drop the line to clear the bank info and keep the provided values. Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events") Link: https://lore.kernel.org/r/20240527151852.29036-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ump_convert.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index c21be87f5da9..f5d22dd00842 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -892,7 +892,6 @@ static int pgm_ev_to_ump_midi2(const struct snd_seq_event *event, data->pg.bank_msb = cc->cc_bank_msb; data->pg.bank_lsb = cc->cc_bank_lsb; cc->bank_set = 0; - cc->cc_bank_msb = cc->cc_bank_lsb = 0; } return 1; } From e662c90a6debc3bd8d8eff916ad21d9ec458dfcd Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 27 May 2024 14:38:08 -0500 Subject: [PATCH 12/26] ALSA/hda: intel-dsp-config: reduce log verbosity The information on PCI class/subclass was interesting in the Skylake timeframe, since the DSP was only enabled on a limited number of platforms. Now most Intel platforms do enable the DSP, so the information is less interesting to log. When a DSP driver is used, the common helper may be called multiple times due to deferred probes, but there's no reason to print the same information multiple times. Using dev_info_once() covers all the existing usages for internal cards with DSPs. External cards don't rely on DSPs so far. Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20240527193808.165652-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index cfdb1b73c88c..537863447358 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -668,7 +668,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) return SND_INTEL_DSP_DRIVER_LEGACY; } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + dev_dbg(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); /* find the configuration for the specific device */ cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table)); @@ -678,12 +678,12 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) if (cfg->flags & FLAG_SOF) { if (cfg->flags & FLAG_SOF_ONLY_IF_SOUNDWIRE && snd_intel_dsp_check_soundwire(pci) > 0) { - dev_info(&pci->dev, "SoundWire enabled on CannonLake+ platform, using SOF driver\n"); + dev_info_once(&pci->dev, "SoundWire enabled on CannonLake+ platform, using SOF driver\n"); return SND_INTEL_DSP_DRIVER_SOF; } if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC && snd_intel_dsp_check_dmic(pci)) { - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + dev_info_once(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); return SND_INTEL_DSP_DRIVER_SOF; } if (!(cfg->flags & FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE)) @@ -694,7 +694,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) if (cfg->flags & FLAG_SST) { if (cfg->flags & FLAG_SST_ONLY_IF_DMIC) { if (snd_intel_dsp_check_dmic(pci)) { - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SST driver\n"); + dev_info_once(&pci->dev, "Digital mics found on Skylake+ platform, using SST driver\n"); return SND_INTEL_DSP_DRIVER_SST; } } else { From edb32776196afa393c074d6a2733e3a69e66b299 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 May 2024 10:37:59 +0200 Subject: [PATCH 13/26] ALSA: seq: Fix incorrect UMP type for system messages When converting a legacy system message to a UMP packet, it forgot to modify the UMP type field but keeping the default type (either type 2 or 4). Correct to the right type for system messages. Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events") Cc: Link: https://lore.kernel.org/r/20240529083800.5742-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ump_convert.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index f5d22dd00842..add70b4c2885 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -739,6 +739,7 @@ static int system_1p_ev_to_ump_midi1(const struct snd_seq_event *event, union snd_ump_midi1_msg *data, unsigned char status) { + data->system.type = UMP_MSG_TYPE_SYSTEM; // override data->system.status = status; data->system.parm1 = event->data.control.value & 0x7f; return 1; @@ -750,6 +751,7 @@ static int system_2p_ev_to_ump_midi1(const struct snd_seq_event *event, union snd_ump_midi1_msg *data, unsigned char status) { + data->system.type = UMP_MSG_TYPE_SYSTEM; // override data->system.status = status; data->system.parm1 = (event->data.control.value >> 7) & 0x7f; data->system.parm2 = event->data.control.value & 0x7f; From fe85f6e607d75b856e7229924c71f55e005f8284 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 May 2024 10:38:21 +0200 Subject: [PATCH 14/26] ALSA: ump: Don't clear bank selection after sending a program change The current code clears the bank selection MSB/LSB after sending a program change, but this can be wrong, as many apps may not send the full bank selection with both MSB and LSB but sending only one. Better to keep the previous bank set. Fixes: 0b5288f5fe63 ("ALSA: ump: Add legacy raw MIDI support") Cc: Link: https://lore.kernel.org/r/20240529083823.5778-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/ump_convert.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/ump_convert.c b/sound/core/ump_convert.c index de04799fdb69..f67c44c83fde 100644 --- a/sound/core/ump_convert.c +++ b/sound/core/ump_convert.c @@ -404,7 +404,6 @@ static int cvt_legacy_cmd_to_ump(struct ump_cvt_to_ump *cvt, midi2->pg.bank_msb = cc->cc_bank_msb; midi2->pg.bank_lsb = cc->cc_bank_lsb; cc->bank_set = 0; - cc->cc_bank_msb = cc->cc_bank_lsb = 0; } break; case UMP_MSG_STATUS_CHANNEL_PRESSURE: From 6d40dbc75877110c5e0d661dd77f6cfce916765e Mon Sep 17 00:00:00 2001 From: Alexandre Belloni Date: Tue, 28 May 2024 21:18:50 +0200 Subject: [PATCH 15/26] ALSA: pcm: fix typo in comment Fix the typo in the comment for SNDRV_PCM_RATE_KNOT Signed-off-by: Alexandre Belloni Link: https://lore.kernel.org/r/20240528191850.63314-1-alexandre.belloni@bootlin.com Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 61c6054618c8..3edd7a7346da 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -124,7 +124,7 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_768000 (1U<<16) /* 768000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1U<<30) /* continuous range */ -#define SNDRV_PCM_RATE_KNOT (1U<<31) /* supports more non-continuos rates */ +#define SNDRV_PCM_RATE_KNOT (1U<<31) /* supports more non-continuous rates */ #define SNDRV_PCM_RATE_8000_44100 (SNDRV_PCM_RATE_8000|SNDRV_PCM_RATE_11025|\ SNDRV_PCM_RATE_16000|SNDRV_PCM_RATE_22050|\ From b062938fd9afec844c50571fddd8d81623a60ee1 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Mon, 27 May 2024 14:19:40 -0500 Subject: [PATCH 16/26] ASoC: Intel: sof-sdw: fix missing SPI_MASTER dependency The addition of the Cirrus Logic 'sidecar' amps adds a dependency on SPI_MASTER. Kconfig warnings: (for reference only) WARNING: unmet direct dependencies detected for SND_SOC_CS35L56_SPI Depends on [n]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SPI_MASTER [=n] && (SOUNDWIRE [=y] || !SOUNDWIRE [=y]) Selected by [y]: - SND_SOC_INTEL_SOUNDWIRE_SOF_MACH [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_SOC_INTEL_MACH [=y] && SND_SOC_SOF_INTEL_SOUNDWIRE [=y] && I2C [=y] && ACPI [=y] && (MFD_INTEL_LPSS [=y] || COMPILE_TEST [=n]) && (SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES [=y] || COMPILE_TEST [=n]) && SOUNDWIRE [=y] Fixes: b831b4dca48d ("ASoC: intel: sof_sdw: Add support for cs42l43-cs35l56 sidecar amps") Reported-by: kernel test robot Closes: https://lore.kernel.org/oe-kbuild-all/202405140758.o2HY4nYD-lkp@intel.com/ Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Link: https://msgid.link/r/20240527191940.30107-1-pierre-louis.bossart@linux.intel.com Reviewed-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/intel/boards/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3ed81ab649c5..4e0586034de4 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -652,7 +652,7 @@ if SND_SOC_SOF_INTEL_SOUNDWIRE config SND_SOC_INTEL_SOUNDWIRE_SOF_MACH tristate "SoundWire generic machine driver" - depends on I2C && ACPI + depends on I2C && SPI_MASTER && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SND_SOC_INTEL_USER_FRIENDLY_LONG_NAMES || COMPILE_TEST depends on SOUNDWIRE From ffa077b2f6ad124ec3d23fbddc5e4b0ff2647af8 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 29 May 2024 15:12:01 +0300 Subject: [PATCH 17/26] ASoC: SOF: ipc4-topology: Fix input format query of process modules without base extension If a process module does not have base config extension then the same format applies to all of it's inputs and the process->base_config_ext is NULL, causing NULL dereference when specifically crafted topology and sequences used. Fixes: 648fea128476 ("ASoC: SOF: ipc4-topology: set copier output format for process module") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Cc: stable@vger.kernel.org Link: https://msgid.link/r/20240529121201.14687-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index beff10989324..33e8c5f7d9ca 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -217,6 +217,14 @@ sof_ipc4_get_input_pin_audio_fmt(struct snd_sof_widget *swidget, int pin_index) } process = swidget->private; + + /* + * For process modules without base config extension, base module config + * format is used for all input pins + */ + if (process->init_config != SOF_IPC4_MODULE_INIT_CONFIG_TYPE_BASE_CFG_WITH_EXT) + return &process->base_config.audio_fmt; + base_cfg_ext = process->base_config_ext; /* From ac0d71ee534e67c7e53439e8e9cb45ed40731660 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 May 2024 18:47:16 +0200 Subject: [PATCH 18/26] ALSA: ump: Don't accept an invalid UMP protocol number When a UMP Stream Configuration message is received, the driver tries to switch the protocol, but there was no sanity check of the protocol, hence it can pass an invalid value. Add the check and bail out if a wrong value is passed. Fixes: a79807683781 ("ALSA: ump: Add helper to change MIDI protocol") Cc: Link: https://lore.kernel.org/r/20240529164723.18309-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/ump.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/core/ump.c b/sound/core/ump.c index fd6a68a54278..117c7ecc4856 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -685,10 +685,17 @@ static void seq_notify_protocol(struct snd_ump_endpoint *ump) */ int snd_ump_switch_protocol(struct snd_ump_endpoint *ump, unsigned int protocol) { + unsigned int type; + protocol &= ump->info.protocol_caps; if (protocol == ump->info.protocol) return 0; + type = protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI_MASK; + if (type != SNDRV_UMP_EP_INFO_PROTO_MIDI1 && + type != SNDRV_UMP_EP_INFO_PROTO_MIDI2) + return 0; + ump->info.protocol = protocol; ump_dbg(ump, "New protocol = %x (caps = %x)\n", protocol, ump->info.protocol_caps); From bc42ca002d5d211f9c57334b9b4c25ddb0b4ec35 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 29 May 2024 18:47:17 +0200 Subject: [PATCH 19/26] ALSA: ump: Set default protocol when not given explicitly When an inquiry of the current protocol via UMP Stream Configuration message fails by some reason, we may leave the current protocol undefined, which may lead to unexpected behavior. Better to assume a valid protocol found in the protocol capability bits instead. For a device that doesn't support the UMP v1.2 feature, it won't reach to this code path, and USB MIDI GTB descriptor would be used for determining the protocol, instead. Link: https://lore.kernel.org/r/20240529164723.18309-2-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/ump.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/core/ump.c b/sound/core/ump.c index 117c7ecc4856..3f61220c23b4 100644 --- a/sound/core/ump.c +++ b/sound/core/ump.c @@ -967,6 +967,14 @@ int snd_ump_parse_endpoint(struct snd_ump_endpoint *ump) if (err < 0) ump_dbg(ump, "Unable to get UMP EP stream config\n"); + /* If no protocol is set by some reason, assume the valid one */ + if (!(ump->info.protocol & SNDRV_UMP_EP_INFO_PROTO_MIDI_MASK)) { + if (ump->info.protocol_caps & SNDRV_UMP_EP_INFO_PROTO_MIDI2) + ump->info.protocol |= SNDRV_UMP_EP_INFO_PROTO_MIDI2; + else if (ump->info.protocol_caps & SNDRV_UMP_EP_INFO_PROTO_MIDI1) + ump->info.protocol |= SNDRV_UMP_EP_INFO_PROTO_MIDI1; + } + /* Query and create blocks from Function Blocks */ for (blk = 0; blk < ump->info.num_blocks; blk++) { err = create_block_from_fb_info(ump, blk); From 700fe6fd093d08c6da2bda8efe00479b0e617327 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 May 2024 12:10:43 +0200 Subject: [PATCH 20/26] ALSA: seq: Fix yet another spot for system message conversion We fixed the incorrect UMP type for system messages in the recent commit, but it missed one place in system_ev_to_ump_midi1(). Fix it now. Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events") Fixes: c2bb79613fed ("ALSA: seq: Fix incorrect UMP type for system messages") Link: https://lore.kernel.org/r/20240530101044.17524-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ump_convert.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index add70b4c2885..f0f332017e66 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -729,6 +729,7 @@ static int system_ev_to_ump_midi1(const struct snd_seq_event *event, union snd_ump_midi1_msg *data, unsigned char status) { + data->system.type = UMP_MSG_TYPE_SYSTEM; // override data->system.status = status; return 1; } From 49cb894d567980235b6e64d5e69950ff77debd8c Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:14 +0300 Subject: [PATCH 21/26] ASoC: SOF: ipc4-topology: Add support for NHLT with 16-bit only DMIC blob The ACPI NHLT table always had 32-bit DMIC blob even if 16-bit was also present and taken as a 'rule' which obviously got broken and there is at least one device on the market which ships with only 16-bit DMIC configuration blob. This corner case has never been supported and it is going to need topology updates for DMIC copier to support multiple formats. As for the kernel side: if the copier supports multiple formats and the preferred 32-bit DMIC blob is not found then we will try to get a 16-bit DMIC configuration and look for a 16-bit copier config. Fixes: f9209644ae76 ("ASoC: SOF: ipc4-topology: Correct DAI copier config and NHLT blob request") Link: https://github.com/thesofproject/linux/issues/4973 Signed-off-by: Peter Ujfalusi Reviewed-by: Seppo Ingalsuo Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240530111918.21974-2-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index beff10989324..521b4dcba601 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1483,6 +1483,8 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai dir, dev_type); if (!cfg) { + bool get_new_blob = false; + if (format_change) { /* * The 32-bit blob was not found in NHLT table, try to @@ -1490,7 +1492,20 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai */ bit_depth = params_width(params); format_change = false; + get_new_blob = true; + } else if (linktype == SOF_DAI_INTEL_DMIC && !single_format) { + /* + * The requested 32-bit blob (no format change for the + * blob request) was not found in NHLT table, try to + * look for 16-bit blob if the copier supports multiple + * formats + */ + bit_depth = 16; + format_change = true; + get_new_blob = true; + } + if (get_new_blob) { cfg = intel_nhlt_get_endpoint_blob(sdev->dev, ipc4_data->nhlt, dai_index, nhlt_type, bit_depth, bit_depth, @@ -1513,8 +1528,8 @@ out: if (format_change) { /* - * Update the params to reflect that we have loaded 32-bit blob - * instead of the 16-bit. + * Update the params to reflect that different blob was loaded + * instead of the requested bit depth (16 -> 32 or 32 -> 16). * This information is going to be used by the caller to find * matching copier format on the dai side. */ @@ -1522,7 +1537,11 @@ out: m = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); snd_mask_none(m); - snd_mask_set_format(m, SNDRV_PCM_FORMAT_S32_LE); + if (bit_depth == 16) + snd_mask_set_format(m, SNDRV_PCM_FORMAT_S16_LE); + else + snd_mask_set_format(m, SNDRV_PCM_FORMAT_S32_LE); + } return 0; From 2a865c9c3fb0289a95f1cb51b42d248736ff45cb Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:15 +0300 Subject: [PATCH 22/26] ASoC: SOF: ipc4-topology: Print out the channel count in sof_ipc4_dbg_audio_format Print out the number of channels for the format explicitly instead of having the reader to understand how to interpret the ch_map and ch_cfg values. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Bard Liao Link: https://msgid.link/r/20240530111918.21974-3-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 521b4dcba601..2e0234b865ad 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -195,9 +195,10 @@ static void sof_ipc4_dbg_audio_format(struct device *dev, struct sof_ipc4_pin_fo for (i = 0; i < num_formats; i++) { struct sof_ipc4_audio_format *fmt = &pin_fmt[i].audio_fmt; dev_dbg(dev, - "Pin index #%d: %uHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x) buffer size %d\n", - pin_fmt[i].pin_index, fmt->sampling_frequency, fmt->bit_depth, fmt->ch_map, - fmt->ch_cfg, fmt->interleaving_style, fmt->fmt_cfg, + "Pin index #%d: %uHz, %ubit, %luch (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x) buffer size %d\n", + pin_fmt[i].pin_index, fmt->sampling_frequency, fmt->bit_depth, + SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg), + fmt->ch_map, fmt->ch_cfg, fmt->interleaving_style, fmt->fmt_cfg, pin_fmt[i].buffer_size); } } From 3b64fd2f83f203f5a34faed3dadf6464313f827d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:16 +0300 Subject: [PATCH 23/26] ASoC: SOF: ipc4-topology/pcm: Rename sof_ipc4_copier_is_single_format() Rename the sof_ipc4_copier_is_single_format() to sof_ipc4_copier_is_single_bitdepth() to clear the confusion of the use of 'format' when we are querying information on the bit depth. Format is used to describe a combination of parameters (rate, channels, sample format / bit depth). Rename the flags used to store the result at the same time. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240530111918.21974-4-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-pcm.c | 12 +++++------ sound/soc/sof/ipc4-topology.c | 40 +++++++++++++++++------------------ sound/soc/sof/ipc4-topology.h | 6 +++--- 3 files changed, 29 insertions(+), 29 deletions(-) diff --git a/sound/soc/sof/ipc4-pcm.c b/sound/soc/sof/ipc4-pcm.c index 307bee63756b..4df2be3d39eb 100644 --- a/sound/soc/sof/ipc4-pcm.c +++ b/sound/soc/sof/ipc4-pcm.c @@ -650,7 +650,7 @@ static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(rtd, 0); struct sof_ipc4_audio_format *ipc4_fmt; struct sof_ipc4_copier *ipc4_copier; - bool single_fmt = false; + bool single_bitdepth = false; u32 valid_bits = 0; int dir, ret; @@ -682,18 +682,18 @@ static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, return 0; if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - if (sof_ipc4_copier_is_single_format(sdev, + if (sof_ipc4_copier_is_single_bitdepth(sdev, available_fmt->output_pin_fmts, available_fmt->num_output_formats)) { ipc4_fmt = &available_fmt->output_pin_fmts->audio_fmt; - single_fmt = true; + single_bitdepth = true; } } else { - if (sof_ipc4_copier_is_single_format(sdev, + if (sof_ipc4_copier_is_single_bitdepth(sdev, available_fmt->input_pin_fmts, available_fmt->num_input_formats)) { ipc4_fmt = &available_fmt->input_pin_fmts->audio_fmt; - single_fmt = true; + single_bitdepth = true; } } } @@ -703,7 +703,7 @@ static int sof_ipc4_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, if (ret) return ret; - if (single_fmt) { + if (single_bitdepth) { snd_mask_none(fmt); valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(ipc4_fmt->fmt_cfg); dev_dbg(component->dev, "Set %s to %d bit format\n", dai->name, valid_bits); diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 2e0234b865ad..afb7908766a3 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1423,7 +1423,7 @@ static int snd_sof_get_hw_config_params(struct snd_sof_dev *sdev, struct snd_sof static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, - bool single_format, + bool single_bitdepth, struct snd_pcm_hw_params *params, u32 dai_index, u32 linktype, u8 dir, u32 **dst, u32 *len) { @@ -1446,7 +1446,7 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai * Look for 32-bit blob first instead of 16-bit if copier * supports multiple formats */ - if (bit_depth == 16 && !single_format) { + if (bit_depth == 16 && !single_bitdepth) { dev_dbg(sdev->dev, "Looking for 32-bit blob first for DMIC\n"); format_change = true; bit_depth = 32; @@ -1494,7 +1494,7 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai bit_depth = params_width(params); format_change = false; get_new_blob = true; - } else if (linktype == SOF_DAI_INTEL_DMIC && !single_format) { + } else if (linktype == SOF_DAI_INTEL_DMIC && !single_bitdepth) { /* * The requested 32-bit blob (no format change for the * blob request) was not found in NHLT table, try to @@ -1550,7 +1550,7 @@ out: #else static int snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, - bool single_format, + bool single_bitdepth, struct snd_pcm_hw_params *params, u32 dai_index, u32 linktype, u8 dir, u32 **dst, u32 *len) { @@ -1558,9 +1558,9 @@ snd_sof_get_nhlt_endpoint_data(struct snd_sof_dev *sdev, struct snd_sof_dai *dai } #endif -bool sof_ipc4_copier_is_single_format(struct snd_sof_dev *sdev, - struct sof_ipc4_pin_format *pin_fmts, - u32 pin_fmts_size) +bool sof_ipc4_copier_is_single_bitdepth(struct snd_sof_dev *sdev, + struct sof_ipc4_pin_format *pin_fmts, + u32 pin_fmts_size) { struct sof_ipc4_audio_format *fmt; u32 valid_bits; @@ -1591,7 +1591,7 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, struct snd_pcm_hw_params dai_params = *params; struct sof_ipc4_copier_data *copier_data; struct sof_ipc4_copier *ipc4_copier; - bool single_format; + bool single_bitdepth; int ret; ipc4_copier = dai->private; @@ -1605,12 +1605,12 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, * format lookup */ if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - single_format = sof_ipc4_copier_is_single_format(sdev, + single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, available_fmt->output_pin_fmts, available_fmt->num_output_formats); /* Update the dai_params with the only supported format */ - if (single_format) { + if (single_bitdepth) { ret = sof_ipc4_update_hw_params(sdev, &dai_params, &available_fmt->output_pin_fmts[0].audio_fmt, BIT(SNDRV_PCM_HW_PARAM_FORMAT)); @@ -1618,12 +1618,12 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, return ret; } } else { - single_format = sof_ipc4_copier_is_single_format(sdev, + single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, available_fmt->input_pin_fmts, available_fmt->num_input_formats); /* Update the dai_params with the only supported format */ - if (single_format) { + if (single_bitdepth) { ret = sof_ipc4_update_hw_params(sdev, &dai_params, &available_fmt->input_pin_fmts[0].audio_fmt, BIT(SNDRV_PCM_HW_PARAM_FORMAT)); @@ -1632,7 +1632,7 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, } } - ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_format, + ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_bitdepth, &dai_params, ipc4_copier->dai_index, ipc4_copier->dai_type, dir, @@ -1667,7 +1667,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, u32 out_ref_rate, out_ref_channels; u32 deep_buffer_dma_ms = 0; int output_fmt_index; - bool single_output_format; + bool single_output_bitdepth; int i; dev_dbg(sdev->dev, "copier %s, type %d", swidget->widget->name, swidget->id); @@ -1804,9 +1804,9 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, return ret; /* set the reference params for output format selection */ - single_output_format = sof_ipc4_copier_is_single_format(sdev, - available_fmt->output_pin_fmts, - available_fmt->num_output_formats); + single_output_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, + available_fmt->output_pin_fmts, + available_fmt->num_output_formats); switch (swidget->id) { case snd_soc_dapm_aif_in: case snd_soc_dapm_dai_out: @@ -1818,7 +1818,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, out_ref_rate = in_fmt->sampling_frequency; out_ref_channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(in_fmt->fmt_cfg); - if (!single_output_format) + if (!single_output_bitdepth) out_ref_valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(in_fmt->fmt_cfg); break; @@ -1827,7 +1827,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, case snd_soc_dapm_dai_in: out_ref_rate = params_rate(fe_params); out_ref_channels = params_channels(fe_params); - if (!single_output_format) { + if (!single_output_bitdepth) { out_ref_valid_bits = sof_ipc4_get_valid_bits(sdev, fe_params); if (out_ref_valid_bits < 0) return out_ref_valid_bits; @@ -1845,7 +1845,7 @@ sof_ipc4_prepare_copier_module(struct snd_sof_widget *swidget, * if the output format is the same across all available output formats, choose * that as the reference. */ - if (single_output_format) { + if (single_output_bitdepth) { struct sof_ipc4_audio_format *out_fmt; out_fmt = &available_fmt->output_pin_fmts[0].audio_fmt; diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 4488762f6a71..f4dc499c0ffe 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -476,7 +476,7 @@ struct sof_ipc4_process { u32 init_config; }; -bool sof_ipc4_copier_is_single_format(struct snd_sof_dev *sdev, - struct sof_ipc4_pin_format *pin_fmts, - u32 pin_fmts_size); +bool sof_ipc4_copier_is_single_bitdepth(struct snd_sof_dev *sdev, + struct sof_ipc4_pin_format *pin_fmts, + u32 pin_fmts_size); #endif From 2fcad03eaba1b86e6b829f73a9e75e681b7f3106 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:17 +0300 Subject: [PATCH 24/26] ASoC: SOF: ipc4-topology: Improve readability of sof_ipc4_prepare_dai_copier() Remove the duplicated code paths to check for single bit depth and to update the params with storing the parameters needed by the function and have a single code section. No functional change but the code is easier to follow. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240530111918.21974-5-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 40 +++++++++++++++-------------------- 1 file changed, 17 insertions(+), 23 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index afb7908766a3..645252789cfe 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1590,8 +1590,10 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, struct sof_ipc4_available_audio_format *available_fmt; struct snd_pcm_hw_params dai_params = *params; struct sof_ipc4_copier_data *copier_data; + struct sof_ipc4_pin_format *pin_fmts; struct sof_ipc4_copier *ipc4_copier; bool single_bitdepth; + u32 num_pin_fmts; int ret; ipc4_copier = dai->private; @@ -1605,31 +1607,23 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, * format lookup */ if (dir == SNDRV_PCM_STREAM_PLAYBACK) { - single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, - available_fmt->output_pin_fmts, - available_fmt->num_output_formats); - - /* Update the dai_params with the only supported format */ - if (single_bitdepth) { - ret = sof_ipc4_update_hw_params(sdev, &dai_params, - &available_fmt->output_pin_fmts[0].audio_fmt, - BIT(SNDRV_PCM_HW_PARAM_FORMAT)); - if (ret) - return ret; - } + pin_fmts = available_fmt->output_pin_fmts; + num_pin_fmts = available_fmt->num_output_formats; } else { - single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, - available_fmt->input_pin_fmts, - available_fmt->num_input_formats); + pin_fmts = available_fmt->input_pin_fmts; + num_pin_fmts = available_fmt->num_input_formats; + } - /* Update the dai_params with the only supported format */ - if (single_bitdepth) { - ret = sof_ipc4_update_hw_params(sdev, &dai_params, - &available_fmt->input_pin_fmts[0].audio_fmt, - BIT(SNDRV_PCM_HW_PARAM_FORMAT)); - if (ret) - return ret; - } + single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, pin_fmts, + num_pin_fmts); + + /* Update the dai_params with the only supported format */ + if (single_bitdepth) { + ret = sof_ipc4_update_hw_params(sdev, &dai_params, + &pin_fmts[0].audio_fmt, + BIT(SNDRV_PCM_HW_PARAM_FORMAT)); + if (ret) + return ret; } ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_bitdepth, From b65456b7b379e20ab225a4e906dc4a0c98fddd7a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 30 May 2024 14:19:18 +0300 Subject: [PATCH 25/26] ASoC: SOF: ipc4-topology: Adjust the params based on DAI formats Currently we only check the bit depth value among to DAI formats, but other parameters might be constant, like number of channels and/or rate. In capture we use the fe params as a reference to find the format and blob which should be used, but in the path we can have components which can handle expanding/narrowing number of channels or do a resample. In these cases the topology is expected to have 'fixed' parameter for channels/rates/bit depth and the conversion to the fe format is going to be done within the path. In practice this patch fixes issues like: All DMIC formats are fixed four channels We have a component which converts the four channel to stereo FE is opened with 2 channel Even if we have the correct bit depth format and blob (for four channel) we will still be looking for stereo configurations, which will fail. Note: the adjustment of params have switched order with the checking of single bit depth (needed for the NHLT blob fallback support). This change is non function, just that if the sof_ipc4_narrow_params_to_format() would fail, there is no point of checking the single bit depth. Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Seppo Ingalsuo Reviewed-by: Ranjani Sridharan Link: https://msgid.link/r/20240530111918.21974-6-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/ipc4-topology.c | 71 ++++++++++++++++++++++++++++------- 1 file changed, 57 insertions(+), 14 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 645252789cfe..1bea4bab958b 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1583,6 +1583,55 @@ bool sof_ipc4_copier_is_single_bitdepth(struct snd_sof_dev *sdev, return true; } +static int +sof_ipc4_adjust_params_to_dai_format(struct snd_sof_dev *sdev, + struct snd_pcm_hw_params *params, + struct sof_ipc4_pin_format *pin_fmts, + u32 pin_fmts_size) +{ + u32 params_mask = BIT(SNDRV_PCM_HW_PARAM_RATE) | + BIT(SNDRV_PCM_HW_PARAM_CHANNELS) | + BIT(SNDRV_PCM_HW_PARAM_FORMAT); + struct sof_ipc4_audio_format *fmt; + u32 rate, channels, valid_bits; + int i; + + fmt = &pin_fmts[0].audio_fmt; + rate = fmt->sampling_frequency; + channels = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg); + valid_bits = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); + + /* check if parameters in topology defined formats are the same */ + for (i = 1; i < pin_fmts_size; i++) { + u32 val; + + fmt = &pin_fmts[i].audio_fmt; + + if (params_mask & BIT(SNDRV_PCM_HW_PARAM_RATE)) { + val = fmt->sampling_frequency; + if (val != rate) + params_mask &= ~BIT(SNDRV_PCM_HW_PARAM_RATE); + } + if (params_mask & BIT(SNDRV_PCM_HW_PARAM_CHANNELS)) { + val = SOF_IPC4_AUDIO_FORMAT_CFG_CHANNELS_COUNT(fmt->fmt_cfg); + if (val != channels) + params_mask &= ~BIT(SNDRV_PCM_HW_PARAM_CHANNELS); + } + if (params_mask & BIT(SNDRV_PCM_HW_PARAM_FORMAT)) { + val = SOF_IPC4_AUDIO_FORMAT_CFG_V_BIT_DEPTH(fmt->fmt_cfg); + if (val != valid_bits) + params_mask &= ~BIT(SNDRV_PCM_HW_PARAM_FORMAT); + } + } + + if (params_mask) + return sof_ipc4_update_hw_params(sdev, params, + &pin_fmts[0].audio_fmt, + params_mask); + + return 0; +} + static int sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, struct snd_pcm_hw_params *params, int dir) @@ -1601,10 +1650,9 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, available_fmt = &ipc4_copier->available_fmt; /* - * If the copier on the DAI side supports only single bit depth then - * this depth (format) should be used to look for the NHLT blob (if - * needed) and in case of capture this should be used for the input - * format lookup + * Fixup the params based on the format parameters of the DAI. If any + * of the RATE, CHANNELS, bit depth is static among the formats then + * narrow the params to only allow that specific parameter value. */ if (dir == SNDRV_PCM_STREAM_PLAYBACK) { pin_fmts = available_fmt->output_pin_fmts; @@ -1614,18 +1662,13 @@ sof_ipc4_prepare_dai_copier(struct snd_sof_dev *sdev, struct snd_sof_dai *dai, num_pin_fmts = available_fmt->num_input_formats; } + ret = sof_ipc4_adjust_params_to_dai_format(sdev, &dai_params, pin_fmts, + num_pin_fmts); + if (ret) + return ret; + single_bitdepth = sof_ipc4_copier_is_single_bitdepth(sdev, pin_fmts, num_pin_fmts); - - /* Update the dai_params with the only supported format */ - if (single_bitdepth) { - ret = sof_ipc4_update_hw_params(sdev, &dai_params, - &pin_fmts[0].audio_fmt, - BIT(SNDRV_PCM_HW_PARAM_FORMAT)); - if (ret) - return ret; - } - ret = snd_sof_get_nhlt_endpoint_data(sdev, dai, single_bitdepth, &dai_params, ipc4_copier->dai_index, From 310fa3ec2859f1c094e6e9b5d2e1ca51738c409a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 31 May 2024 09:51:07 +0200 Subject: [PATCH 26/26] ALSA: seq: ump: Fix swapped song position pointer data At converting between the legacy event and UMP, the parameters for MIDI Song Position Pointer are incorrectly stored. It should have been LSB -> MSB order while it stored in MSB -> LSB order. This patch corrects the ordering. Fixes: e9e02819a98a ("ALSA: seq: Automatic conversion of UMP events") Link: https://lore.kernel.org/r/20240531075110.3250-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/core/seq/seq_ump_convert.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/seq/seq_ump_convert.c b/sound/core/seq/seq_ump_convert.c index f0f332017e66..171fb75267af 100644 --- a/sound/core/seq/seq_ump_convert.c +++ b/sound/core/seq/seq_ump_convert.c @@ -157,7 +157,7 @@ static void ump_system_to_one_param_ev(const union snd_ump_midi1_msg *val, static void ump_system_to_songpos_ev(const union snd_ump_midi1_msg *val, struct snd_seq_event *ev) { - ev->data.control.value = (val->system.parm1 << 7) | val->system.parm2; + ev->data.control.value = (val->system.parm2 << 7) | val->system.parm1; } /* Encoders for 0xf0 - 0xff */ @@ -754,8 +754,8 @@ static int system_2p_ev_to_ump_midi1(const struct snd_seq_event *event, { data->system.type = UMP_MSG_TYPE_SYSTEM; // override data->system.status = status; - data->system.parm1 = (event->data.control.value >> 7) & 0x7f; - data->system.parm2 = event->data.control.value & 0x7f; + data->system.parm1 = event->data.control.value & 0x7f; + data->system.parm2 = (event->data.control.value >> 7) & 0x7f; return 1; }