diff options
Diffstat (limited to 'sound')
96 files changed, 885 insertions, 522 deletions
diff --git a/sound/core/oss/mulaw.c b/sound/core/oss/mulaw.c index 3788906421a7..fe27034f2846 100644 --- a/sound/core/oss/mulaw.c +++ b/sound/core/oss/mulaw.c @@ -329,8 +329,8 @@ int snd_pcm_plugin_build_mulaw(struct snd_pcm_substream *plug, snd_BUG(); return -EINVAL; } - if (snd_BUG_ON(!snd_pcm_format_linear(format->format))) - return -ENXIO; + if (!snd_pcm_format_linear(format->format)) + return -EINVAL; err = snd_pcm_plugin_build(plug, "Mu-Law<->linear conversion", src_format, dst_format, diff --git a/sound/core/timer.c b/sound/core/timer.c index d9f85f2d66a3..6e27d87b18ed 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -816,9 +816,9 @@ static void snd_timer_clear_callbacks(struct snd_timer *timer, * timer tasklet * */ -static void snd_timer_tasklet(unsigned long arg) +static void snd_timer_tasklet(struct tasklet_struct *t) { - struct snd_timer *timer = (struct snd_timer *) arg; + struct snd_timer *timer = from_tasklet(timer, t, task_queue); unsigned long flags; if (timer->card && timer->card->shutdown) { @@ -967,8 +967,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid, INIT_LIST_HEAD(&timer->ack_list_head); INIT_LIST_HEAD(&timer->sack_list_head); spin_lock_init(&timer->lock); - tasklet_init(&timer->task_queue, snd_timer_tasklet, - (unsigned long)timer); + tasklet_setup(&timer->task_queue, snd_timer_tasklet); timer->max_instances = 1000; /* default limit per timer */ if (card != NULL) { timer->module = card->module; diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f8586f75441d..ee1c428b1fd3 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -64,7 +64,7 @@ #define IT_PKT_HEADER_SIZE_CIP 8 // For 2 CIP header. #define IT_PKT_HEADER_SIZE_NO_CIP 0 // Nothing. -static void pcm_period_tasklet(unsigned long data); +static void pcm_period_tasklet(struct tasklet_struct *t); /** * amdtp_stream_init - initialize an AMDTP stream structure @@ -94,7 +94,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit, s->flags = flags; s->context = ERR_PTR(-1); mutex_init(&s->mutex); - tasklet_init(&s->period_tasklet, pcm_period_tasklet, (unsigned long)s); + tasklet_setup(&s->period_tasklet, pcm_period_tasklet); s->packet_index = 0; init_waitqueue_head(&s->callback_wait); @@ -441,9 +441,9 @@ static void update_pcm_pointers(struct amdtp_stream *s, } } -static void pcm_period_tasklet(unsigned long data) +static void pcm_period_tasklet(struct tasklet_struct *t) { - struct amdtp_stream *s = (void *)data; + struct amdtp_stream *s = from_tasklet(s, t, period_tasklet); struct snd_pcm_substream *pcm = READ_ONCE(s->pcm); if (pcm) diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c index c84b913a9fe0..ab8408966ec3 100644 --- a/sound/firewire/digi00x/digi00x.c +++ b/sound/firewire/digi00x/digi00x.c @@ -14,6 +14,7 @@ MODULE_LICENSE("GPL v2"); #define VENDOR_DIGIDESIGN 0x00a07e #define MODEL_CONSOLE 0x000001 #define MODEL_RACK 0x000002 +#define SPEC_VERSION 0x000001 static int name_card(struct snd_dg00x *dg00x) { @@ -175,14 +176,18 @@ static const struct ieee1394_device_id snd_dg00x_id_table[] = { /* Both of 002/003 use the same ID. */ { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_CONSOLE, }, { .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_VERSION | IEEE1394_MATCH_MODEL_ID, .vendor_id = VENDOR_DIGIDESIGN, + .version = SPEC_VERSION, .model_id = MODEL_RACK, }, {} diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c index 5dac0d9fc58e..75f2edd8e78f 100644 --- a/sound/firewire/tascam/tascam.c +++ b/sound/firewire/tascam/tascam.c @@ -39,9 +39,6 @@ static const struct snd_tscm_spec model_specs[] = { .midi_capture_ports = 2, .midi_playback_ports = 4, }, - // This kernel module doesn't support FE-8 because the most of features - // can be implemented in userspace without any specific support of this - // module. }; static int identify_model(struct snd_tscm *tscm) @@ -211,11 +208,39 @@ static void snd_tscm_remove(struct fw_unit *unit) } static const struct ieee1394_device_id snd_tscm_id_table[] = { + // Tascam, FW-1884. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800000, + }, + // Tascam, FE-8 (.version = 0x800001) + // This kernel module doesn't support FE-8 because the most of features + // can be implemented in userspace without any specific support of this + // module. + // + // .version = 0x800002 is unknown. + // + // Tascam, FW-1082. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = 0x00022e, + .specifier_id = 0x00022e, + .version = 0x800003, + }, + // Tascam, FW-1804. { .match_flags = IEEE1394_MATCH_VENDOR_ID | - IEEE1394_MATCH_SPECIFIER_ID, + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, .vendor_id = 0x00022e, .specifier_id = 0x00022e, + .version = 0x800004, }, {} }; diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c index 09ddab5f5cae..9766f6af8743 100644 --- a/sound/hda/hdac_bus.c +++ b/sound/hda/hdac_bus.c @@ -46,6 +46,18 @@ int snd_hdac_bus_init(struct hdac_bus *bus, struct device *dev, INIT_LIST_HEAD(&bus->hlink_list); init_waitqueue_head(&bus->rirb_wq); bus->irq = -1; + + /* + * Default value of '8' is as per the HD audio specification (Rev 1.0a). + * Following relation is used to derive STRIPE control value. + * For sample rate <= 48K: + * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } + * For sample rate > 48K: + * { ((num_channels * bits_per_sample * rate/48000) / + * number of SDOs) >= 8 } + */ + bus->sdo_limit = 8; + return 0; } EXPORT_SYMBOL_GPL(snd_hdac_bus_init); diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index 011b17cc1efa..b98449fd92f3 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -529,17 +529,6 @@ bool snd_hdac_bus_init_chip(struct hdac_bus *bus, bool full_reset) bus->chip_init = true; - /* - * Default value of '8' is as per the HD audio specification (Rev 1.0a). - * Following relation is used to derive STRIPE control value. - * For sample rate <= 48K: - * { ((num_channels * bits_per_sample) / number of SDOs) >= 8 } - * For sample rate > 48K: - * { ((num_channels * bits_per_sample * rate/48000) / - * number of SDOs) >= 8 } - */ - bus->sdo_limit = 8; - return true; } EXPORT_SYMBOL_GPL(snd_hdac_bus_init_chip); diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 333220f0f8af..3e9e9ac804f6 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -127,6 +127,8 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_init); void snd_hdac_device_exit(struct hdac_device *codec) { pm_runtime_put_noidle(&codec->dev); + /* keep balance of runtime PM child_count in parent device */ + pm_runtime_set_suspended(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); kfree(codec->vendor_name); kfree(codec->chip_name); diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 99aec7349167..1c5114dedda9 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -54,7 +54,7 @@ static const struct config_entry config_table[] = { #endif /* * Apollolake (Broxton-P) - * the legacy HDaudio driver is used except on Up Squared (SOF) and + * the legacy HDAudio driver is used except on Up Squared (SOF) and * Chromebooks (SST) */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) @@ -89,7 +89,7 @@ static const struct config_entry config_table[] = { }, #endif /* - * Skylake and Kabylake use legacy HDaudio driver except for Google + * Skylake and Kabylake use legacy HDAudio driver except for Google * Chromebooks (SST) */ @@ -135,7 +135,7 @@ static const struct config_entry config_table[] = { #endif /* - * Geminilake uses legacy HDaudio driver except for Google + * Geminilake uses legacy HDAudio driver except for Google * Chromebooks */ /* Geminilake */ @@ -157,7 +157,7 @@ static const struct config_entry config_table[] = { /* * CoffeeLake, CannonLake, CometLake, IceLake, TigerLake use legacy - * HDaudio driver except for Google Chromebooks and when DMICs are + * HDAudio driver except for Google Chromebooks and when DMICs are * present. Two cases are required since Coreboot does not expose NHLT * tables. * @@ -391,7 +391,7 @@ int snd_intel_dsp_driver_probe(struct pci_dev *pci) if (pci->class == 0x040300) return SND_INTEL_DSP_DRIVER_LEGACY; if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDAudio legacy driver\n", pci->class); return SND_INTEL_DSP_DRIVER_LEGACY; } diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 5363d88cc4b9..2e5a5c5279e8 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -308,7 +308,7 @@ static inline int verify_mpu401(const struct snd_mpu401 *mpu) } /* - * This is apparently the standard way to initailise an MPU-401 + * This is apparently the standard way to initialise an MPU-401 */ static inline void initialise_mpu401(const struct snd_mpu401 *mpu) { @@ -339,7 +339,7 @@ static void soundscape_free(struct snd_card *c) } /* - * Tell the SoundScape to begin a DMA tranfer using the given channel. + * Tell the SoundScape to begin a DMA transfer using the given channel. * All locking issues are left to the caller. */ static void sscape_start_dma_unsafe(unsigned io_base, enum GA_REG reg) @@ -803,7 +803,7 @@ static int mpu401_open(struct snd_mpu401 *mpu) } /* - * Initialse an MPU-401 subdevice for MIDI support on the SoundScape. + * Initialise an MPU-401 subdevice for MIDI support on the SoundScape. */ static int create_mpu401(struct snd_card *card, int devnum, unsigned long port, int irq) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 023c35a2a951..35e76480306e 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -921,10 +921,10 @@ static void snd_card_asihpi_timer_function(struct timer_list *t) add_timer(&dpcm->timer); } -static void snd_card_asihpi_int_task(unsigned long data) +static void snd_card_asihpi_int_task(struct tasklet_struct *t) { - struct hpi_adapter *a = (struct hpi_adapter *)data; - struct snd_card_asihpi *asihpi; + struct snd_card_asihpi *asihpi = from_tasklet(asihpi, t, t); + struct hpi_adapter *a = asihpi->hpi; WARN_ON(!a || !a->snd_card || !a->snd_card->private_data); asihpi = (struct snd_card_asihpi *)a->snd_card->private_data; @@ -2871,8 +2871,7 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev, if (hpi->interrupt_mode) { asihpi->pcm_start = snd_card_asihpi_pcm_int_start; asihpi->pcm_stop = snd_card_asihpi_pcm_int_stop; - tasklet_init(&asihpi->t, snd_card_asihpi_int_task, - (unsigned long)hpi); + tasklet_setup(&asihpi->t, snd_card_asihpi_int_task); hpi->interrupt_callback = snd_card_asihpi_isr; } else { asihpi->pcm_start = snd_card_asihpi_pcm_timer_start; diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 496dcde9715d..9790f5108a16 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -343,7 +343,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, struct hpi_message hm; struct hpi_response hr; struct hpi_adapter adapter; - struct hpi_pci pci; + struct hpi_pci pci = { 0 }; memset(&adapter, 0, sizeof(adapter)); @@ -499,7 +499,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, return 0; err: - for (idx = 0; idx < HPI_MAX_ADAPTER_MEM_SPACES; idx++) { + while (--idx >= 0) { if (pci.ap_mem_base[idx]) { iounmap(pci.ap_mem_base[idx]); pci.ap_mem_base[idx] = NULL; diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 70d775ff967e..c189f70c82cb 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -537,7 +537,8 @@ static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id, else /* Power down */ chip->spi_dac_reg[reg] |= bit; - return snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]); + if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0) + return -ENXIO; } return 0; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e34a4d5d047c..36a9dbc33aa0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2127,9 +2127,10 @@ static int azx_probe(struct pci_dev *pci, */ if (dmic_detect) { err = snd_intel_dsp_driver_probe(pci); - if (err != SND_INTEL_DSP_DRIVER_ANY && - err != SND_INTEL_DSP_DRIVER_LEGACY) + if (err != SND_INTEL_DSP_DRIVER_ANY && err != SND_INTEL_DSP_DRIVER_LEGACY) { + dev_dbg(&pci->dev, "HDAudio driver not selected, aborting probe\n"); return -ENODEV; + } } else { dev_warn(&pci->dev, "dmic_detect option is deprecated, pass snd-intel-dspcfg.dsp_driver=1 option instead\n"); } @@ -2745,8 +2746,6 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, /* Zhaoxin */ { PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, - /* Loongson */ - { PCI_DEVICE(0x0014, 0x7a07), .driver_data = AZX_DRIVER_GENERIC }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index c94553bcca88..70164d1428d4 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -179,6 +179,10 @@ static int __maybe_unused hda_tegra_runtime_suspend(struct device *dev) struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); if (chip && chip->running) { + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + azx_stop_chip(chip); azx_enter_link_reset(chip); } @@ -200,6 +204,9 @@ static int __maybe_unused hda_tegra_runtime_resume(struct device *dev) if (chip && chip->running) { hda_tegra_init(hda); azx_init_chip(chip, 1); + /* disable controller wake up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); } return 0; diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index b8c8490e568b..402050088090 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2794,6 +2794,7 @@ static void i915_pin_cvt_fixup(struct hda_codec *codec, hda_nid_t cvt_nid) { if (per_pin) { + haswell_verify_D0(codec, per_pin->cvt_nid, per_pin->pin_nid); snd_hda_set_dev_select(codec, per_pin->pin_nid, per_pin->dev_id); intel_verify_pin_cvt_connect(codec, per_pin); @@ -3734,6 +3735,7 @@ static int tegra_hdmi_build_pcms(struct hda_codec *codec) static int patch_tegra_hdmi(struct hda_codec *codec) { + struct hdmi_spec *spec; int err; err = patch_generic_hdmi(codec); @@ -3741,6 +3743,10 @@ static int patch_tegra_hdmi(struct hda_codec *codec) return err; codec->patch_ops.build_pcms = tegra_hdmi_build_pcms; + spec = codec->spec; + spec->chmap.ops.chmap_cea_alloc_validate_get_type = + nvhdmi_chmap_cea_alloc_validate_get_type; + spec->chmap.ops.chmap_validate = nvhdmi_chmap_validate; return 0; } @@ -4263,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI", patch_i915_glk_hdmi), HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7f9d35273734..d4f17b465892 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3427,7 +3427,11 @@ static void alc256_shutup(struct hda_codec *codec) /* 3k pull low control for Headset jack. */ /* NOTE: call this before clearing the pin, otherwise codec stalls */ - alc_update_coef_idx(codec, 0x46, 0, 3 << 12); + /* If disable 3k pulldown control for alc257, the Mic detection will not work correctly + * when booting with headset plugged. So skip setting it for the codec alc257 + */ + if (codec->core.vendor_id != 0x10ec0257) + alc_update_coef_idx(codec, 0x46, 0, 3 << 12); if (!spec->no_shutup_pins) snd_hda_codec_write(codec, hp_pin, 0, @@ -5867,6 +5871,39 @@ static void alc275_fixup_gpio4_off(struct hda_codec *codec, } } +/* Quirk for Thinkpad X1 7th and 8th Gen + * The following fixed routing needed + * DAC1 (NID 0x02) -> Speaker (NID 0x14); some eq applied secretly + * DAC2 (NID 0x03) -> Bass (NID 0x17) & Headphone (NID 0x21); sharing a DAC + * DAC3 (NID 0x06) -> Unused, due to the lack of volume amp + */ +static void alc285_fixup_thinkpad_x1_gen7(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const hda_nid_t conn[] = { 0x02, 0x03 }; /* exclude 0x06 */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, 0x17, 0x03, 0x21, 0x03, 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + case HDA_FIXUP_ACT_BUILD: + /* The generic parser creates somewhat unintuitive volume ctls + * with the fixed routing above, and the shared DAC2 may be + * confusing for PA. + * Rename those to unique names so that PA doesn't touch them + * and use only Master volume. + */ + rename_ctl(codec, "Front Playback Volume", "DAC1 Playback Volume"); + rename_ctl(codec, "Bass Speaker Playback Volume", "DAC2 Playback Volume"); + break; + } +} + static void alc233_alc662_fixup_lenovo_dual_codecs(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5959,6 +5996,40 @@ static void alc_fixup_disable_mic_vref(struct hda_codec *codec, snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); } + +static void alc294_gx502_toggle_output(struct hda_codec *codec, + struct hda_jack_callback *cb) +{ + /* The Windows driver sets the codec up in a very different way where + * it appears to leave 0x10 = 0x8a20 set. For Linux we need to toggle it + */ + if (snd_hda_jack_detect_state(codec, 0x21) == HDA_JACK_PRESENT) + alc_write_coef_idx(codec, 0x10, 0x8a20); + else + alc_write_coef_idx(codec, 0x10, 0x0a20); +} + +static void alc294_fixup_gx502_hp(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* Pin 0x21: headphones/headset mic */ + if (!is_jack_detectable(codec, 0x21)) + return; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_jack_detect_enable_callback(codec, 0x21, + alc294_gx502_toggle_output); + break; + case HDA_FIXUP_ACT_INIT: + /* Make sure to start in a correct state, i.e. if + * headphones have been plugged in before powering up the system + */ + alc294_gx502_toggle_output(codec, NULL); + break; + } +} + static void alc285_fixup_hp_gpio_amp_init(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -5983,6 +6054,7 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec, #include "hp_x360_helper.c" enum { + ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, ALC269_FIXUP_DELL_M101Z, @@ -6135,9 +6207,13 @@ enum { ALC289_FIXUP_DUAL_SPK, ALC294_FIXUP_SPK2_TO_DAC1, ALC294_FIXUP_ASUS_DUAL_SPK, + ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, + ALC294_FIXUP_ASUS_GX502_HP, + ALC294_FIXUP_ASUS_GX502_PINS, + ALC294_FIXUP_ASUS_GX502_VERBS, ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED, @@ -6156,9 +6232,14 @@ enum { ALC269_FIXUP_LEMOTE_A1802, ALC269_FIXUP_LEMOTE_A190X, ALC256_FIXUP_INTEL_NUC8_RUGGED, + ALC255_FIXUP_XIAOMI_HEADSET_MIC, }; static const struct hda_fixup alc269_fixups[] = { + [ALC269_FIXUP_GPIO2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_gpio2, + }, [ALC269_FIXUP_SONY_VAIO] = { .type = HDA_FIXUP_PINCTLS, .v.pins = (const struct hda_pintbl[]) { @@ -6978,6 +7059,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC233_FIXUP_LENOVO_MULTI_CODECS] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_alc662_fixup_lenovo_dual_codecs, + .chained = true, + .chain_id = ALC269_FIXUP_GPIO2 }, [ALC233_FIXUP_ACER_HEADSET_MIC] = { .type = HDA_FIXUP_VERBS, @@ -7280,11 +7363,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, + [ALC285_FIXUP_THINKPAD_X1_GEN7] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_thinkpad_x1_gen7, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI + }, [ALC285_FIXUP_THINKPAD_HEADSET_JACK] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_jack, .chained = true, - .chain_id = ALC285_FIXUP_SPEAKER2_TO_DAC1 + .chain_id = ALC285_FIXUP_THINKPAD_X1_GEN7 }, [ALC294_FIXUP_ASUS_HPE] = { .type = HDA_FIXUP_VERBS, @@ -7297,6 +7386,33 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_ASUS_HEADSET_MIC }, + [ALC294_FIXUP_ASUS_GX502_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, /* front HP mic */ + { 0x1a, 0x01a11830 }, /* rear external mic */ + { 0x21, 0x03211020 }, /* front HP out */ + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_VERBS + }, + [ALC294_FIXUP_ASUS_GX502_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* set 0x15 to HP-OUT ctrl */ + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, + /* unmute the 0x15 amp */ + { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_GX502_HP + }, + [ALC294_FIXUP_ASUS_GX502_HP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc294_fixup_gx502_hp, + }, [ALC294_FIXUP_ASUS_COEF_1B] = { .type = HDA_FIXUP_VERBS, .v.verbs = (const struct hda_verb[]) { @@ -7486,6 +7602,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC255_FIXUP_XIAOMI_HEADSET_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, + { } + }, + .chained = true, + .chain_id = ALC289_FIXUP_ASUS_GA401 + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7670,6 +7796,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1f11, "ASUS Zephyrus G14", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x3030, "ASUS ZN270IE", ALC256_FIXUP_ASUS_AIO_GPIO2), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS S101", ALC269_FIXUP_STEREO_DMIC), @@ -7694,6 +7821,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x144d, 0xc169, "Samsung Notebook 9 Pen (NP930SBE-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc176, "Samsung Notebook 9 Pro (NP930MBE-K04US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc189, "Samsung Galaxy Flex Book (NT950QCG-X716)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc18a, "Samsung Galaxy Book Ion (NP930XCJ-K01US)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8), SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), @@ -7779,6 +7909,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b35, 0x1236, "CZC TMI", ALC269_FIXUP_CZC_TMI), SND_PCI_QUIRK(0x1b35, 0x1237, "CZC L101", ALC269_FIXUP_CZC_L101), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), @@ -7955,6 +8086,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, {.id = ALC298_FIXUP_HUAWEI_MBX_STEREO, .name = "huawei-mbx-stereo"}, {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, + {.id = ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, .name = "alc298-samsung-headphone"}, + {.id = ALC255_FIXUP_XIAOMI_HEADSET_MIC, .name = "alc255-xiaomi-headset"}, {} }; #define ALC225_STANDARD_PINS \ diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b4f300281822..098c69b3b7aa 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1070,9 +1070,9 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval, return 0; } -static void riptide_handleirq(unsigned long dev_id) +static void riptide_handleirq(struct tasklet_struct *t) { - struct snd_riptide *chip = (void *)dev_id; + struct snd_riptide *chip = from_tasklet(chip, t, riptide_tq); struct cmdif *cif = chip->cif; struct snd_pcm_substream *substream[PLAYBACK_SUBSTREAMS + 1]; struct snd_pcm_runtime *runtime; @@ -1843,7 +1843,7 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci, chip->received_irqs = 0; chip->handled_irqs = 0; chip->cif = NULL; - tasklet_init(&chip->riptide_tq, riptide_handleirq, (unsigned long)chip); + tasklet_setup(&chip->riptide_tq, riptide_handleirq); if ((chip->res_port = request_region(chip->port, 64, "RIPTIDE")) == NULL) { diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 227aece17e39..dda56ecfd33b 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -3791,9 +3791,9 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) return 0; } -static void hdsp_midi_tasklet(unsigned long arg) +static void hdsp_midi_tasklet(struct tasklet_struct *t) { - struct hdsp *hdsp = (struct hdsp *)arg; + struct hdsp *hdsp = from_tasklet(hdsp, t, midi_tasklet); if (hdsp->midi[0].pending) snd_hdsp_midi_input_read (&hdsp->midi[0]); @@ -5182,7 +5182,7 @@ static int snd_hdsp_create(struct snd_card *card, spin_lock_init(&hdsp->lock); - tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp); + tasklet_setup(&hdsp->midi_tasklet, hdsp_midi_tasklet); pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev); hdsp->firmware_rev &= 0xff; diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 0fa49f4d15cf..572350aaf18d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2169,9 +2169,9 @@ static int snd_hdspm_create_midi(struct snd_card *card, } -static void hdspm_midi_tasklet(unsigned long arg) +static void hdspm_midi_tasklet(struct tasklet_struct *t) { - struct hdspm *hdspm = (struct hdspm *)arg; + struct hdspm *hdspm = from_tasklet(hdspm, t, midi_tasklet); int i = 0; while (i < hdspm->midiPorts) { @@ -6836,8 +6836,7 @@ static int snd_hdspm_create(struct snd_card *card, } - tasklet_init(&hdspm->midi_tasklet, - hdspm_midi_tasklet, (unsigned long) hdspm); + tasklet_setup(&hdspm->midi_tasklet, hdspm_midi_tasklet); if (hdspm->io_type != MADIface) { diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index b8161a08f2ca..58bb49fff184 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -227,14 +227,14 @@ static int snd_ps3_program_dma(struct snd_ps3_card_info *card, switch (filltype) { case SND_PS3_DMA_FILLTYPE_SILENT_FIRSTFILL: silent = 1; - /* intentionally fall thru */ + fallthrough; case SND_PS3_DMA_FILLTYPE_FIRSTFILL: ch0_kick_event = PS3_AUDIO_KICK_EVENT_ALWAYS; break; case SND_PS3_DMA_FILLTYPE_SILENT_RUNNING: silent = 1; - /* intentionally fall thru */ + fallthrough; case SND_PS3_DMA_FILLTYPE_RUNNING: ch0_kick_event = PS3_AUDIO_KICK_EVENT_SERIALOUT0_EMPTY; break; diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 55815fdaa1aa..406526e79af3 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -138,7 +138,7 @@ static int acp3x_1015_hw_params(struct snd_pcm_substream *substream, srate = params_rate(params); for_each_rtd_codec_dais(rtd, i, codec_dai) { - if (strcmp(codec_dai->component->name, "rt1015-aif")) + if (strcmp(codec_dai->name, "rt1015-aif")) continue; ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); if (ret < 0) diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 623dfd3ea705..7b14d9a81b97 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -314,40 +314,30 @@ static int acp_pdm_dma_close(struct snd_soc_component *component, return 0; } -static int acp_pdm_dai_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) +static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) { struct pdm_stream_instance *rtd; + int ret; + bool pdm_status; unsigned int ch_mask; rtd = substream->runtime->private_data; - switch (params_channels(params)) { + ret = 0; + switch (substream->runtime->channels) { case TWO_CH: ch_mask = 0x00; break; default: return -EINVAL; } - rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); - rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + - ACP_WOV_PDM_DECIMATION_FACTOR); - return 0; -} - -static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct pdm_stream_instance *rtd; - int ret; - bool pdm_status; - - rtd = substream->runtime->private_data; - ret = 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + rn_writel(ch_mask, rtd->acp_base + ACP_WOV_PDM_NO_OF_CHANNELS); + rn_writel(PDM_DECIMATION_FACTOR, rtd->acp_base + + ACP_WOV_PDM_DECIMATION_FACTOR); rtd->bytescount = acp_pdm_get_byte_count(rtd, substream->stream); pdm_status = check_pdm_dma_status(rtd->acp_base); @@ -369,7 +359,6 @@ static int acp_pdm_dai_trigger(struct snd_pcm_substream *substream, } static struct snd_soc_dai_ops acp_pdm_dai_ops = { - .hw_params = acp_pdm_dai_hw_params, .trigger = acp_pdm_dai_trigger, }; diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 3cb63886195f..04acc18f2d72 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -536,7 +536,7 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, /* cpu is BCLK master */ mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT; set_divs = 1; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_CBM_CFM: /* cpu is slave */ mra |= MCHP_I2SMCC_MRA_MODE_SLAVE; diff --git a/sound/soc/codecs/jz4770.c b/sound/soc/codecs/jz4770.c index c0a28f06b09a..298689a07168 100644 --- a/sound/soc/codecs/jz4770.c +++ b/sound/soc/codecs/jz4770.c @@ -202,7 +202,7 @@ static int jz4770_codec_set_bias_level(struct snd_soc_component *codec, REG_CR_VIC_SB_SLEEP, REG_CR_VIC_SB_SLEEP); regmap_update_bits(regmap, JZ4770_CODEC_REG_CR_VIC, REG_CR_VIC_SB, REG_CR_VIC_SB); - /* fall-through */ + fallthrough; default: break; } diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index 5fe724728e84..e4675cfff7b2 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -838,8 +838,8 @@ static int max98373_sdw_probe(struct sdw_slave *slave, /* Regmap Initialization */ regmap = devm_regmap_init_sdw(slave, &max98373_sdw_regmap); - if (!regmap) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); return max98373_init(slave, regmap); } diff --git a/sound/soc/codecs/msm8916-wcd-analog.c b/sound/soc/codecs/msm8916-wcd-analog.c index 4428c62e25cf..3ddd822240e3 100644 --- a/sound/soc/codecs/msm8916-wcd-analog.c +++ b/sound/soc/codecs/msm8916-wcd-analog.c @@ -19,8 +19,8 @@ #define CDC_D_REVISION1 (0xf000) #define CDC_D_PERPH_SUBTYPE (0xf005) -#define CDC_D_INT_EN_SET (0x015) -#define CDC_D_INT_EN_CLR (0x016) +#define CDC_D_INT_EN_SET (0xf015) +#define CDC_D_INT_EN_CLR (0xf016) #define MBHC_SWITCH_INT BIT(7) #define MBHC_MIC_ELECTRICAL_INS_REM_DET BIT(6) #define MBHC_BUTTON_PRESS_DET BIT(5) diff --git a/sound/soc/codecs/pcm186x.c b/sound/soc/codecs/pcm186x.c index f0da55901dcb..b8845f45549e 100644 --- a/sound/soc/codecs/pcm186x.c +++ b/sound/soc/codecs/pcm186x.c @@ -401,7 +401,7 @@ static int pcm186x_set_fmt(struct snd_soc_dai *dai, unsigned int format) break; case SND_SOC_DAIFMT_DSP_A: priv->tdm_offset += 1; - /* fall through */ + fallthrough; /* DSP_A uses the same basic config as DSP_B * except we need to shift the TDM output by one BCK cycle */ diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 5e445fee4ef5..821e7395f90f 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -306,6 +306,13 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(dai->component); int ret; + /* + * Some sound card sets 0 Hz as reset, + * but it is impossible to set. Ignore it here + */ + if (freq == 0) + return 0; + if (freq > PCM3168A_MAX_SYSCLK) return -EINVAL; diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index b0ba0d2acbdd..56e952a904a3 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -684,8 +684,8 @@ static int rt1308_sdw_probe(struct sdw_slave *slave, /* Regmap Initialization */ regmap = devm_regmap_init_sdw(slave, &rt1308_sdw_regmap); - if (!regmap) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); rt1308_sdw_init(&slave->dev, regmap, slave); diff --git a/sound/soc/codecs/rt700-sdw.c b/sound/soc/codecs/rt700-sdw.c index 4d14048d1197..1d24bf040718 100644 --- a/sound/soc/codecs/rt700-sdw.c +++ b/sound/soc/codecs/rt700-sdw.c @@ -452,8 +452,8 @@ static int rt700_sdw_probe(struct sdw_slave *slave, /* Regmap Initialization */ sdw_regmap = devm_regmap_init_sdw(slave, &rt700_sdw_regmap); - if (!sdw_regmap) - return -EINVAL; + if (IS_ERR(sdw_regmap)) + return PTR_ERR(sdw_regmap); regmap = devm_regmap_init(&slave->dev, NULL, &slave->dev, &rt700_regmap); diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 45b928954b58..7efff130a638 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -452,8 +452,8 @@ static int rt711_sdw_probe(struct sdw_slave *slave, /* Regmap Initialization */ sdw_regmap = devm_regmap_init_sdw(slave, &rt711_sdw_regmap); - if (!sdw_regmap) - return -EINVAL; + if (IS_ERR(sdw_regmap)) + return PTR_ERR(sdw_regmap); regmap = devm_regmap_init(&slave->dev, NULL, &slave->dev, &rt711_regmap); diff --git a/sound/soc/codecs/rt715-sdw.c b/sound/soc/codecs/rt715-sdw.c index d11b23d6b240..68a36739f1b0 100644 --- a/sound/soc/codecs/rt715-sdw.c +++ b/sound/soc/codecs/rt715-sdw.c @@ -527,8 +527,8 @@ static int rt715_sdw_probe(struct sdw_slave *slave, /* Regmap Initialization */ sdw_regmap = devm_regmap_init_sdw(slave, &rt715_sdw_regmap); - if (!sdw_regmap) - return -EINVAL; + if (IS_ERR(sdw_regmap)) + return PTR_ERR(sdw_regmap); regmap = devm_regmap_init(&slave->dev, NULL, &slave->dev, &rt715_regmap); diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 5cd50d841177..8efe20605f9b 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -842,6 +842,18 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) goto out; + if (adcx140->supply_areg == NULL) + sleep_cfg_val |= ADCX140_AREG_INTERNAL; + + ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val); + if (ret) { + dev_err(adcx140->dev, "setting sleep config failed %d\n", ret); + goto out; + } + + /* 8.4.3: Wait >= 1ms after entering active mode. */ + usleep_range(1000, 100000); + pdm_count = device_property_count_u32(adcx140->dev, "ti,pdm-edge-select"); if (pdm_count <= ADCX140_NUM_PDM_EDGES && pdm_count > 0) { @@ -889,18 +901,6 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) goto out; - if (adcx140->supply_areg == NULL) - sleep_cfg_val |= ADCX140_AREG_INTERNAL; - - ret = regmap_write(adcx140->regmap, ADCX140_SLEEP_CFG, sleep_cfg_val); - if (ret) { - dev_err(adcx140->dev, "setting sleep config failed %d\n", ret); - goto out; - } - - /* 8.4.3: Wait >= 1ms after entering active mode. */ - usleep_range(1000, 100000); - ret = regmap_update_bits(adcx140->regmap, ADCX140_BIAS_CFG, ADCX140_MIC_BIAS_VAL_MSK | ADCX140_MIC_BIAS_VREF_MSK, bias_cfg); @@ -980,6 +980,8 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, if (!adcx140) return -ENOMEM; + adcx140->dev = &i2c->dev; + adcx140->gpio_reset = devm_gpiod_get_optional(adcx140->dev, "reset", GPIOD_OUT_LOW); if (IS_ERR(adcx140->gpio_reset)) @@ -1007,7 +1009,7 @@ static int adcx140_i2c_probe(struct i2c_client *i2c, ret); return ret; } - adcx140->dev = &i2c->dev; + i2c_set_clientdata(i2c, adcx140); return devm_snd_soc_register_component(&i2c->dev, diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 68a3b48e6b31..3bce9a14f0f3 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -412,8 +412,12 @@ int wm8958_aif_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct wm8994 *control = dev_get_drvdata(component->dev->parent); int i; + if (control->type != WM8958) + return 0; + switch (event) { case SND_SOC_DAPM_POST_PMU: case SND_SOC_DAPM_PRE_PMU: diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 317916cb4e27..0623a2251084 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -151,7 +151,6 @@ static const struct reg_default wm8962_reg[] = { { 40, 0x0000 }, /* R40 - SPKOUTL volume */ { 41, 0x0000 }, /* R41 - SPKOUTR volume */ - { 48, 0x0000 }, /* R48 - Additional control(4) */ { 49, 0x0010 }, /* R49 - Class D Control 1 */ { 51, 0x0003 }, /* R51 - Class D Control 2 */ @@ -842,6 +841,7 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_SPKOUTL_VOLUME: case WM8962_SPKOUTR_VOLUME: case WM8962_THERMAL_SHUTDOWN_STATUS: + case WM8962_ADDITIONAL_CONTROL_4: case WM8962_CLASS_D_CONTROL_1: case WM8962_CLASS_D_CONTROL_2: case WM8962_CLOCKING_4: diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a84ae879d37e..fc9ea198ac79 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -43,10 +43,12 @@ #define WM8994_NUM_DRC 3 #define WM8994_NUM_EQ 3 -static struct { +struct wm8994_reg_mask { unsigned int reg; unsigned int mask; -} wm8994_vu_bits[] = { +}; + +static struct wm8994_reg_mask wm8994_vu_bits[] = { { WM8994_LEFT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_RIGHT_LINE_INPUT_1_2_VOLUME, WM8994_IN1_VU }, { WM8994_LEFT_LINE_INPUT_3_4_VOLUME, WM8994_IN2_VU }, @@ -60,14 +62,10 @@ static struct { { WM8994_AIF1_DAC1_LEFT_VOLUME, WM8994_AIF1DAC1_VU }, { WM8994_AIF1_DAC1_RIGHT_VOLUME, WM8994_AIF1DAC1_VU }, - { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, - { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, { WM8994_AIF2_DAC_LEFT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF2_DAC_RIGHT_VOLUME, WM8994_AIF2DAC_VU }, { WM8994_AIF1_ADC1_LEFT_VOLUME, WM8994_AIF1ADC1_VU }, { WM8994_AIF1_ADC1_RIGHT_VOLUME, WM8994_AIF1ADC1_VU }, - { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, - { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_AIF2_ADC_LEFT_VOLUME, WM8994_AIF2ADC_VU }, { WM8994_AIF2_ADC_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, { WM8994_DAC1_LEFT_VOLUME, WM8994_DAC1_VU }, @@ -76,6 +74,14 @@ static struct { { WM8994_DAC2_RIGHT_VOLUME, WM8994_DAC2_VU }, }; +/* VU bitfields for ADC2, DAC2 not available on WM1811 */ +static struct wm8994_reg_mask wm8994_adc2_dac2_vu_bits[] = { + { WM8994_AIF1_DAC2_LEFT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_DAC2_RIGHT_VOLUME, WM8994_AIF1DAC2_VU }, + { WM8994_AIF1_ADC2_LEFT_VOLUME, WM8994_AIF1ADC2_VU }, + { WM8994_AIF1_ADC2_RIGHT_VOLUME, WM8994_AIF1ADC2_VU }, +}; + static int wm8994_drc_base[] = { WM8994_AIF1_DRC1_1, WM8994_AIF1_DRC2_1, @@ -1030,6 +1036,26 @@ static bool wm8994_check_class_w_digital(struct snd_soc_component *component) return true; } +static void wm8994_update_vu_bits(struct snd_soc_component *component) +{ + struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); + struct wm8994 *control = wm8994->wm8994; + int i; + + for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) + snd_soc_component_write(component, wm8994_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_vu_bits[i].reg)); + if (control->type == WM1811) + return; + + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_write(component, + wm8994_adc2_dac2_vu_bits[i].reg, + snd_soc_component_read(component, + wm8994_adc2_dac2_vu_bits[i].reg)); +} + static int aif_mclk_set(struct snd_soc_component *component, int aif, bool enable) { struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); @@ -1076,7 +1102,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, struct wm8994_priv *wm8994 = snd_soc_component_get_drvdata(component); struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; - int ret, i; + int ret; int dac; int adc; int val; @@ -1144,10 +1170,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -1181,7 +1204,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - int ret, i; + int ret; int dac; int adc; int val; @@ -1237,10 +1260,7 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_POST_PMU: - for (i = 0; i < ARRAY_SIZE(wm8994_vu_bits); i++) - snd_soc_component_write(component, wm8994_vu_bits[i].reg, - snd_soc_component_read(component, - wm8994_vu_bits[i].reg)); + wm8994_update_vu_bits(component); break; case SND_SOC_DAPM_PRE_PMD: @@ -3494,6 +3514,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * return -EINVAL; } + pm_runtime_get_sync(component->dev); + switch (micbias) { case 1: micdet = &wm8994->micdet[0]; @@ -3541,6 +3563,8 @@ int wm8994_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * snd_soc_dapm_sync(dapm); + pm_runtime_put(component->dev); + return 0; } EXPORT_SYMBOL_GPL(wm8994_mic_detect); @@ -3912,6 +3936,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * return -EINVAL; } + pm_runtime_get_sync(component->dev); + if (jack) { snd_soc_dapm_force_enable_pin(dapm, "CLK_SYS"); snd_soc_dapm_sync(dapm); @@ -3980,6 +4006,8 @@ int wm8958_mic_detect(struct snd_soc_component *component, struct snd_soc_jack * snd_soc_dapm_sync(dapm); } + pm_runtime_put(component->dev); + return 0; } EXPORT_SYMBOL_GPL(wm8958_mic_detect); @@ -4173,11 +4201,13 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994->hubs.dcs_readback_mode = 2; break; } + wm8994->hubs.micd_scthr = true; break; case WM8958: wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.micd_scthr = true; switch (control->revision) { case 0: @@ -4346,6 +4376,14 @@ static int wm8994_component_probe(struct snd_soc_component *component) wm8994_vu_bits[i].mask, wm8994_vu_bits[i].mask); + if (control->type != WM1811) { + for (i = 0; i < ARRAY_SIZE(wm8994_adc2_dac2_vu_bits); i++) + snd_soc_component_update_bits(component, + wm8994_adc2_dac2_vu_bits[i].reg, + wm8994_adc2_dac2_vu_bits[i].mask, + wm8994_adc2_dac2_vu_bits[i].mask); + } + /* Set the low bit of the 3D stereo depth so TLV matches */ snd_soc_component_update_bits(component, WM8994_AIF1_DAC1_FILTERS_2, 1 << WM8994_AIF1DAC1_3D_GAIN_SHIFT, diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 891effe220fe..0c881846f485 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1223,6 +1223,9 @@ int wm_hubs_handle_analogue_pdata(struct snd_soc_component *component, snd_soc_component_update_bits(component, WM8993_ADDITIONAL_CONTROL, WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB); + if (!hubs->micd_scthr) + return 0; + snd_soc_component_update_bits(component, WM8993_MICBIAS, WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK | WM8993_MICB1_LVL | WM8993_MICB2_LVL, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 4b8e5f0d6e32..988b29e63060 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -27,6 +27,7 @@ struct wm_hubs_data { int hp_startup_mode; int series_startup; int no_series_update; + bool micd_scthr; bool no_cache_dac_hp_direct; struct list_head dcs_cache; diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de136c0a497d..52adedc03245 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -73,6 +73,7 @@ struct cpu_priv { * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure + * @streams: Mask of current active streams * @sample_rate: Current sample rate * @sample_format: Current sample format * @asrc_rate: ASRC sample rate used by Back-Ends @@ -89,6 +90,7 @@ struct fsl_asoc_card_priv { struct codec_priv codec_priv; struct cpu_priv cpu_priv; struct snd_soc_card card; + u8 streams; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; @@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct codec_priv *codec_priv = &priv->codec_priv; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; + unsigned int pll_out; int ret; priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); + priv->streams |= BIT(substream->stream); - /* - * If codec-dai is DAI Master and all configurations are already in the - * set_bias_level(), bypass the remaining settings in hw_params(). - * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. - */ - if ((priv->card.set_bias_level && - priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || - fsl_asoc_card_is_ac97(priv)) + if (fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -174,7 +172,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->sysclk_dir[tx]); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set sysclk for cpu dai\n"); - return ret; + goto fail; } if (cpu_priv->slot_width) { @@ -182,6 +180,68 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, cpu_priv->slot_width); if (ret && ret != -ENOTSUPP) { dev_err(dev, "failed to set TDM slot for cpu dai\n"); + goto fail; + } + } + + /* Specific configuration for PLL */ + if (codec_priv->pll_id && codec_priv->fll_id) { + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + goto fail; + } + + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + goto fail; + } + } + + return 0; + +fail: + priv->streams &= ~BIT(substream->stream); + return ret; +} + +static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->streams &= ~BIT(substream->stream); + + if (!priv->streams && codec_priv->pll_id && codec_priv->fll_id) { + /* Force freq to be 0 to avoid error message in codec */ + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->mclk_id, + 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, 0, 0, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to stop FLL: %d\n", ret); return ret; } } @@ -191,6 +251,7 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, static const struct snd_soc_ops fsl_asoc_card_ops = { .hw_params = fsl_asoc_card_hw_params, + .hw_free = fsl_asoc_card_hw_free, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -254,75 +315,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = { }, }; -static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct codec_priv *codec_priv = &priv->codec_priv; - struct device *dev = card->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = asoc_rtd_to_codec(rtd, 0); - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level != SND_SOC_BIAS_STANDBY) - break; - - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, - codec_priv->mclk_id, - codec_priv->mclk_freq, pll_out); - if (ret) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, - pll_out, SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level != SND_SOC_BIAS_PREPARE) - break; - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, - codec_priv->mclk_freq, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); - if (ret) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - return 0; -} - static int fsl_asoc_card_audmux_init(struct device_node *np, struct fsl_asoc_card_priv *priv) { @@ -611,7 +603,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; - priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; @@ -628,26 +619,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_AC97; priv->card.dapm_routes = audio_map_ac97; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { codec_dai_name = "fsl-mqs-dai"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF; @@ -657,7 +644,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { codec_dai_name = "wm8524-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 4ae36099ae82..79b861afd986 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -708,9 +708,9 @@ static void fsl_esai_trigger_stop(struct fsl_esai *esai_priv, bool tx) ESAI_xFCR_xFR, 0); } -static void fsl_esai_hw_reset(unsigned long arg) +static void fsl_esai_hw_reset(struct tasklet_struct *t) { - struct fsl_esai *esai_priv = (struct fsl_esai *)arg; + struct fsl_esai *esai_priv = from_tasklet(esai_priv, t, task); bool tx = true, rx = false, enabled[2]; unsigned long lock_flags; u32 tfcr, rfcr; @@ -1070,8 +1070,7 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } - tasklet_init(&esai_priv->task, fsl_esai_hw_reset, - (unsigned long)esai_priv); + tasklet_setup(&esai_priv->task, fsl_esai_hw_reset); pm_runtime_enable(&pdev->dev); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index d8b9c6547142..404be27c15fe 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -898,7 +898,7 @@ static int _fsl_ssi_set_dai_fmt(struct fsl_ssi *ssi, unsigned int fmt) "missing baudclk for master mode\n"); return -EINVAL; } - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_CBM_CFS: ssi->i2s_net |= SSI_SCR_I2S_MODE_MASTER; break; diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 9e4f66b6b92b..231984882176 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -339,7 +339,6 @@ static int psc_dma_new(struct snd_soc_component *component, static void psc_dma_free(struct snd_soc_component *component, struct snd_pcm *pcm) { - struct snd_soc_pcm_runtime *rtd = pcm->private_data; struct snd_pcm_substream *substream; int stream; diff --git a/sound/soc/hisilicon/hi6210-i2s.c b/sound/soc/hisilicon/hi6210-i2s.c index fd5dcd6b9f85..907f5f1f7b44 100644 --- a/sound/soc/hisilicon/hi6210-i2s.c +++ b/sound/soc/hisilicon/hi6210-i2s.c @@ -261,13 +261,13 @@ static int hi6210_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_U16_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fall through */ + fallthrough; case SNDRV_PCM_FORMAT_S16_LE: bits = HII2S_BITS_16; break; case SNDRV_PCM_FORMAT_U24_LE: signed_data = HII2S_I2S_CFG__S2_CODEC_DATA_FORMAT; - /* fall through */ + fallthrough; case SNDRV_PCM_FORMAT_S24_LE: bits = HII2S_BITS_24; break; diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 49b9f18472bc..fba2c795ce0d 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -331,7 +331,18 @@ static int sst_media_open(struct snd_pcm_substream *substream, ret_val = power_up_sst(stream); if (ret_val < 0) - return ret_val; + goto out_power_up; + + /* + * Make sure the period to be multiple of 1ms to align the + * design of firmware. Apply same rule to buffer size to make + * sure alsa could always find a value for period size + * regardless the buffer size given by user space. + */ + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 48); + snd_pcm_hw_constraint_step(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_SIZE, 48); /* Make sure, that the period size is always even */ snd_pcm_hw_constraint_step(substream->runtime, 0, @@ -340,8 +351,9 @@ static int sst_media_open(struct snd_pcm_substream *substream, return snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); out_ops: - kfree(stream); mutex_unlock(&sst_lock); +out_power_up: + kfree(stream); return ret_val; } diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 54a66cc6db89..d2cda33b65d5 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -181,7 +181,7 @@ static int sst_byt_pcm_trigger(struct snd_soc_component *component, break; case SNDRV_PCM_TRIGGER_SUSPEND: pdata->restore_stream = false; - /* fallthrough */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 414ae4bb5224..7ae34b49815c 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -573,7 +573,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) break; default: dev_err(dev, "get speaker GPIO failed: %d\n", ret); - /* fall through */ + fallthrough; case -EPROBE_DEFER: return ret; } diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 479992f4e97a..fc202747ba83 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -591,6 +591,16 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { /* MPMAN Converter 9, similar hw as the I.T.Works TW891 2-in-1 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "MPMAN"), + DMI_MATCH(DMI_PRODUCT_NAME, "Converter9"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_MONO_SPEAKER | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { /* MPMAN MPWIN895CL */ .matches = { diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 4e2897596cea..688b5e0a49e3 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -1009,7 +1009,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) default: dev_err(&pdev->dev, "Failed to get ext-amp-enable GPIO: %d\n", ret_val); - /* fall through */ + fallthrough; case -EPROBE_DEFER: put_device(codec_dev); return ret_val; @@ -1029,7 +1029,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) default: dev_err(&pdev->dev, "Failed to get hp-detect GPIO: %d\n", ret_val); - /* fall through */ + fallthrough; case -EPROBE_DEFER: put_device(codec_dev); return ret_val; diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index ca4900036ead..bc50eda297ab 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -181,7 +181,7 @@ static void skl_set_hda_codec_autosuspend_delay(struct snd_soc_card *card) struct snd_soc_dai *dai; for_each_card_rtds(card, rtd) { - if (!strstr(rtd->dai_link->codecs->name, "ehdaudio")) + if (!strstr(rtd->dai_link->codecs->name, "ehdaudio0D0")) continue; dai = asoc_rtd_to_codec(rtd, 0); hda_pvt = snd_soc_component_get_drvdata(dai->component); diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index 1a6961592029..b6e63ea13d64 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -66,6 +66,10 @@ int max98373_trigger(struct snd_pcm_substream *substream, int cmd) int j; int ret = 0; + /* set spk pin by playback only */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + for_each_rtd_codec_dais(rtd, j, codec_dai) { struct snd_soc_component *component = codec_dai->component; struct snd_soc_dapm_context *dapm = @@ -86,9 +90,6 @@ int max98373_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - /* Make sure no streams are active before disable pin */ - if (snd_soc_dai_active(codec_dai) != 1) - break; ret = snd_soc_dapm_disable_pin(dapm, pin_name); if (!ret) snd_soc_dapm_sync(dapm); diff --git a/sound/soc/intel/haswell/sst-haswell-dsp.c b/sound/soc/intel/haswell/sst-haswell-dsp.c index de80e19454c1..88c3f63bded9 100644 --- a/sound/soc/intel/haswell/sst-haswell-dsp.c +++ b/sound/soc/intel/haswell/sst-haswell-dsp.c @@ -243,92 +243,45 @@ static irqreturn_t hsw_irq(int irq, void *context) return ret; } -#define CSR_DEFAULT_VALUE 0x8480040E -#define ISC_DEFAULT_VALUE 0x0 -#define ISD_DEFAULT_VALUE 0x0 -#define IMC_DEFAULT_VALUE 0x7FFF0003 -#define IMD_DEFAULT_VALUE 0x7FFF0003 -#define IPCC_DEFAULT_VALUE 0x0 -#define IPCD_DEFAULT_VALUE 0x0 -#define CLKCTL_DEFAULT_VALUE 0x7FF -#define CSR2_DEFAULT_VALUE 0x0 -#define LTR_CTRL_DEFAULT_VALUE 0x0 -#define HMD_CTRL_DEFAULT_VALUE 0x0 - -static void hsw_set_shim_defaults(struct sst_dsp *sst) -{ - sst_dsp_shim_write_unlocked(sst, SST_CSR, CSR_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_ISRX, ISC_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_ISRD, ISD_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IMRX, IMC_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IMRD, IMD_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IPCX, IPCC_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_IPCD, IPCD_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_CLKCTL, CLKCTL_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_CSR2, CSR2_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_LTRC, LTR_CTRL_DEFAULT_VALUE); - sst_dsp_shim_write_unlocked(sst, SST_HMDC, HMD_CTRL_DEFAULT_VALUE); -} - -/* all clock-gating minus DCLCGE and DTCGE */ -#define SST_VDRTCL2_CG_OTHER 0xB7D - static void hsw_set_dsp_D3(struct sst_dsp *sst) { + u32 val; u32 reg; - /* disable clock core gating */ + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* stall, reset and set 24MHz XOSC */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST, - SST_CSR_24MHZ_LPCS | SST_CSR_STALL | SST_CSR_RST); - - /* DRAM power gating all */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg |= SST_VDRTCL0_ISRAMPGE_MASK | - SST_VDRTCL0_DSRAMPGE_MASK; - reg &= ~(SST_VDRTCL0_D3SRAMPGD); - reg |= SST_VDRTCL0_D3PGD; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - udelay(50); + /* enable power gating and switch off DRAM & IRAM blocks */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + val |= SST_VDRTCL0_DSRAMPGE_MASK | + SST_VDRTCL0_ISRAMPGE_MASK; + val &= ~(SST_VDRTCL0_D3PGD | SST_VDRTCL0_D3SRAMPGD); + writel(val, sst->addr.pci_cfg + SST_VDRTCTL0); - /* PLL shutdown enable */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_APLLSE_MASK; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* switch off audio PLL */ + val = readl(sst->addr.pci_cfg + SST_VDRTCTL2); + val |= SST_VDRTCL2_APLLSE_MASK; + writel(val, sst->addr.pci_cfg + SST_VDRTCTL2); - /* disable MCLK */ + /* disable MCLK(clkctl.smos = 0) */ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK, 0); - - /* switch clock gating */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_CG_OTHER; - reg &= ~(SST_VDRTCL2_DTCGE); - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* enable DTCGE separatelly */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DTCGE; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + SST_CLKCTL_MASK, 0); - /* set shim defaults */ - hsw_set_shim_defaults(sst); - - /* set D3 */ - reg = readl(sst->addr.pci_cfg + SST_PMCS); - reg |= SST_PMCS_PS_MASK; - writel(reg, sst->addr.pci_cfg + SST_PMCS); + /* Set D3 state, delay 50 us */ + val = readl(sst->addr.pci_cfg + SST_PMCS); + val |= SST_PMCS_PS_MASK; + writel(val, sst->addr.pci_cfg + SST_PMCS); udelay(50); - /* enable clock core gating */ + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE; + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + udelay(50); + } static void hsw_reset(struct sst_dsp *sst) @@ -346,62 +299,75 @@ static void hsw_reset(struct sst_dsp *sst) SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); } -/* recommended CSR state for power-up */ -#define SST_CSR_D0_MASK (0x18A09C0C | SST_CSR_DCS_MASK) - static int hsw_set_dsp_D0(struct sst_dsp *sst) { - u32 reg; + int tries = 10; + u32 reg, fw_dump_bit; - /* disable clock core gating */ + /* Disable core clock gating (VDRTCTL2.DCLCGE = 0) */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_DCLCGE); + reg &= ~(SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE); writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* switch clock gating */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_CG_OTHER; - reg &= ~(SST_VDRTCL2_DTCGE); - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); + /* Disable D3PG (VDRTCTL0.D3PGD = 1) */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_D3PGD; + writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - /* set D0 */ + /* Set D0 state */ reg = readl(sst->addr.pci_cfg + SST_PMCS); - reg &= ~(SST_PMCS_PS_MASK); + reg &= ~SST_PMCS_PS_MASK; writel(reg, sst->addr.pci_cfg + SST_PMCS); - /* DRAM power gating none */ - reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); - reg &= ~(SST_VDRTCL0_ISRAMPGE_MASK | - SST_VDRTCL0_DSRAMPGE_MASK); - reg |= SST_VDRTCL0_D3SRAMPGD; - reg |= SST_VDRTCL0_D3PGD; - writel(reg, sst->addr.pci_cfg + SST_VDRTCTL0); - mdelay(10); + /* check that ADSP shim is enabled */ + while (tries--) { + reg = readl(sst->addr.pci_cfg + SST_PMCS) & SST_PMCS_PS_MASK; + if (reg == 0) + goto finish; + + msleep(1); + } + + return -ENODEV; - /* set shim defaults */ - hsw_set_shim_defaults(sst); +finish: + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); + + /* stall DSP core, set clk to 192/96Mhz */ + sst_dsp_shim_update_bits_unlocked(sst, + SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK, + SST_CSR_STALL | SST_CSR_DCS(4)); - /* restore MCLK */ + /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, - SST_CLKCTL_MASK, SST_CLKCTL_MASK); + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); - /* PLL shutdown disable */ + /* Stall and reset core, set CSR */ + hsw_reset(sst); + + /* Enable core clock gating (VDRTCTL2.DCLCGE = 1), delay 50 us */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg &= ~(SST_VDRTCL2_APLLSE_MASK); + reg |= SST_VDRTCL2_DCLCGE | SST_VDRTCL2_DTCGE; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, - SST_CSR_D0_MASK, SST_CSR_SBCS0 | SST_CSR_SBCS1 | - SST_CSR_STALL | SST_CSR_DCS(4)); udelay(50); - /* enable clock core gating */ + /* switch on audio PLL */ reg = readl(sst->addr.pci_cfg + SST_VDRTCTL2); - reg |= SST_VDRTCL2_DCLCGE; + reg &= ~SST_VDRTCL2_APLLSE_MASK; writel(reg, sst->addr.pci_cfg + SST_VDRTCTL2); - /* clear reset */ - sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_RST, 0); + /* set default power gating control, enable power gating control for all blocks. that is, + can't be accessed, please enable each block before accessing. */ + reg = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + reg |= SST_VDRTCL0_DSRAMPGE_MASK | SST_VDRTCL0_ISRAMPGE_MASK; + /* for D0, always enable the block(DSRAM[0]) used for FW dump */ + fw_dump_bit = 1 << SST_VDRTCL0_DSRAMPGE_SHIFT; + writel(reg & ~fw_dump_bit, sst->addr.pci_cfg + SST_VDRTCTL0); + /* disable DMA finish function for SSP0 & SSP1 */ sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, @@ -418,6 +384,12 @@ static int hsw_set_dsp_D0(struct sst_dsp *sst) sst_dsp_shim_update_bits(sst, SST_IMRD, (SST_IMRD_DONE | SST_IMRD_BUSY | SST_IMRD_SSP0 | SST_IMRD_DMAC), 0x0); + /* clear IPC registers */ + sst_dsp_shim_write(sst, SST_IPCX, 0x0); + sst_dsp_shim_write(sst, SST_IPCD, 0x0); + sst_dsp_shim_write(sst, 0x80, 0x6); + sst_dsp_shim_write(sst, 0xe0, 0x300a); + return 0; } @@ -443,6 +415,11 @@ static void hsw_sleep(struct sst_dsp *sst) { dev_dbg(sst->dev, "HSW_PM dsp runtime suspend\n"); + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits(sst, SST_CSR, + SST_CSR_24MHZ_LPCS | SST_CSR_RST | SST_CSR_STALL, + SST_CSR_RST | SST_CSR_STALL | SST_CSR_24MHZ_LPCS); + hsw_set_dsp_D3(sst); dev_dbg(sst->dev, "HSW_PM dsp runtime suspend exit\n"); } diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 5dee55e9546b..bbe8d782e0af 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -488,7 +488,7 @@ static int skl_pcm_trigger(struct snd_pcm_substream *substream, int cmd, stream->lpib); snd_hdac_ext_stream_set_lpib(stream, stream->lpib); } - /* fall through */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 36df30915378..c8664ab80d45 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -58,17 +58,17 @@ int axg_tdm_set_tdm_slots(struct snd_soc_dai *dai, u32 *tx_mask, switch (slot_width) { case 0: slot_width = 32; - /* Fall-through */ + fallthrough; case 32: fmt |= SNDRV_PCM_FMTBIT_S32_LE; - /* Fall-through */ + fallthrough; case 24: fmt |= SNDRV_PCM_FMTBIT_S24_LE; fmt |= SNDRV_PCM_FMTBIT_S20_LE; - /* Fall-through */ + fallthrough; case 16: fmt |= SNDRV_PCM_FMTBIT_S16_LE; - /* Fall-through */ + fallthrough; case 8: fmt |= SNDRV_PCM_FMTBIT_S8; break; @@ -133,7 +133,7 @@ static int axg_tdm_iface_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_CBS_CFM: case SND_SOC_DAIFMT_CBM_CFS: dev_err(dai->dev, "only CBS_CFS and CBM_CFM are supported\n"); - /* Fall-through */ + fallthrough; default: return -EINVAL; } diff --git a/sound/soc/meson/axg-toddr.c b/sound/soc/meson/axg-toddr.c index e711abcf8c12..d6adf7edea41 100644 --- a/sound/soc/meson/axg-toddr.c +++ b/sound/soc/meson/axg-toddr.c @@ -18,6 +18,7 @@ #define CTRL0_TODDR_SEL_RESAMPLE BIT(30) #define CTRL0_TODDR_EXT_SIGNED BIT(29) #define CTRL0_TODDR_PP_MODE BIT(28) +#define CTRL0_TODDR_SYNC_CH BIT(27) #define CTRL0_TODDR_TYPE_MASK GENMASK(15, 13) #define CTRL0_TODDR_TYPE(x) ((x) << 13) #define CTRL0_TODDR_MSB_POS_MASK GENMASK(12, 8) @@ -189,10 +190,31 @@ static const struct axg_fifo_match_data axg_toddr_match_data = { .dai_drv = &axg_toddr_dai_drv }; +static int g12a_toddr_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct axg_fifo *fifo = snd_soc_dai_get_drvdata(dai); + int ret; + + ret = axg_toddr_dai_startup(substream, dai); + if (ret) + return ret; + + /* + * Make sure the first channel ends up in the at beginning of the output + * As weird as it looks, without this the first channel may be misplaced + * in memory, with a random shift of 2 channels. + */ + regmap_update_bits(fifo->map, FIFO_CTRL0, CTRL0_TODDR_SYNC_CH, + CTRL0_TODDR_SYNC_CH); + + return 0; +} + static const struct snd_soc_dai_ops g12a_toddr_ops = { .prepare = g12a_toddr_dai_prepare, .hw_params = axg_toddr_dai_hw_params, - .startup = axg_toddr_dai_startup, + .startup = g12a_toddr_dai_startup, .shutdown = axg_toddr_dai_shutdown, }; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d1e09ade0190..c4e7307a4437 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -488,7 +488,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 083413abc2f6..575e2aefefe3 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -143,6 +143,7 @@ static int apq8016_sbc_platform_probe(struct platform_device *pdev) card = &data->card; card->dev = dev; + card->owner = THIS_MODULE; card->dapm_widgets = apq8016_sbc_dapm_widgets; card->num_dapm_widgets = ARRAY_SIZE(apq8016_sbc_dapm_widgets); diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 253549600c5a..1a69baefc5ce 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -114,6 +114,7 @@ static int apq8096_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = dev; + card->owner = THIS_MODULE; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 5194d90ddb96..fd69cf8b1f23 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -52,8 +52,10 @@ int qcom_snd_parse_of(struct snd_soc_card *card) for_each_child_of_node(dev->of_node, np) { dlc = devm_kzalloc(dev, 2 * sizeof(*dlc), GFP_KERNEL); - if (!dlc) - return -ENOMEM; + if (!dlc) { + ret = -ENOMEM; + goto err; + } link->cpus = &dlc[0]; link->platforms = &dlc[1]; diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 2a5302f1db98..0168af849272 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1150,206 +1150,206 @@ static int q6afe_of_xlate_dai_name(struct snd_soc_component *component, } static const struct snd_soc_dapm_widget q6afe_dai_widgets[] = { - SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, 0, 0, 0), + SND_SOC_DAPM_AIF_IN("HDMI_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_0_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_1_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_2_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_3_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_4_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_5_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SLIMBUS_6_RX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_0_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_1_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_2_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_3_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_4_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_5_TX", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SLIMBUS_6_TX", NULL, 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_MI2S_RX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_MI2S_TX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_MI2S_RX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_MI2S_TX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_MI2S_TX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_MI2S_RX_SD1", "Secondary MI2S Playback SD1", - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRI_MI2S_RX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRI_MI2S_TX", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("PRIMARY_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("PRIMARY_TDM_TX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("SEC_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("SEC_TDM_TX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("TERT_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("TERT_TDM_TX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUAT_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUAT_TDM_TX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("QUIN_TDM_RX_7", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_0", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_1", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_2", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_3", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_4", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_5", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_6", NULL, - 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("QUIN_TDM_TX_7", NULL, - 0, 0, 0, 0), - SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, 0, 0, 0), + 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DISPLAY_PORT_RX", "NULL", 0, SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_component_driver q6afe_dai_component = { diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index eaa95b5a7b66..25d23e0266c7 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -973,6 +973,20 @@ static int msm_routing_probe(struct snd_soc_component *c) return 0; } +static unsigned int q6routing_reg_read(struct snd_soc_component *component, + unsigned int reg) +{ + /* default value */ + return 0; +} + +static int q6routing_reg_write(struct snd_soc_component *component, + unsigned int reg, unsigned int val) +{ + /* dummy */ + return 0; +} + static const struct snd_soc_component_driver msm_soc_routing_component = { .probe = msm_routing_probe, .name = DRV_NAME, @@ -981,6 +995,8 @@ static const struct snd_soc_component_driver msm_soc_routing_component = { .num_dapm_widgets = ARRAY_SIZE(msm_qdsp6_widgets), .dapm_routes = intercon, .num_dapm_routes = ARRAY_SIZE(intercon), + .read = q6routing_reg_read, + .write = q6routing_reg_write, }; static int q6pcm_routing_probe(struct platform_device *pdev) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 0d10fba53945..ab1bf23c21a6 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -555,6 +555,7 @@ static int sdm845_snd_platform_probe(struct platform_device *pdev) card->dapm_widgets = sdm845_snd_widgets; card->num_dapm_widgets = ARRAY_SIZE(sdm845_snd_widgets); card->dev = dev; + card->owner = THIS_MODULE; dev_set_drvdata(dev, card); ret = qcom_snd_parse_of(card); if (ret) diff --git a/sound/soc/qcom/storm.c b/sound/soc/qcom/storm.c index c0c388d4db82..80c9cf2f254a 100644 --- a/sound/soc/qcom/storm.c +++ b/sound/soc/qcom/storm.c @@ -96,6 +96,7 @@ static int storm_platform_probe(struct platform_device *pdev) return -ENOMEM; card->dev = &pdev->dev; + card->owner = THIS_MODULE; ret = snd_soc_of_parse_card_name(card, "qcom,model"); if (ret) { diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index 1707414cfa92..5adb293d0435 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -229,13 +229,13 @@ static int rockchip_pdm_hw_params(struct snd_pcm_substream *substream, switch (params_channels(params)) { case 8: val |= PDM_PATH3_EN; - /* fallthrough */ + fallthrough; case 6: val |= PDM_PATH2_EN; - /* fallthrough */ + fallthrough; case 4: val |= PDM_PATH1_EN; - /* fallthrough */ + fallthrough; case 2: val |= PDM_PATH0_EN; break; diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 80ecb5c7fed0..df53d4ea808f 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -733,7 +733,7 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, switch (params_channels(params)) { case 6: val |= MOD_DC2_EN; - /* Fall through */ + fallthrough; case 4: val |= MOD_DC1_EN; break; diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index bd9de77c35f3..50fc7810723e 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -198,9 +198,9 @@ static int siu_pcm_rd_set(struct siu_port *port_info, return 0; } -static void siu_io_tasklet(unsigned long data) +static void siu_io_tasklet(struct tasklet_struct *t) { - struct siu_stream *siu_stream = (struct siu_stream *)data; + struct siu_stream *siu_stream = from_tasklet(siu_stream, t, tasklet); struct snd_pcm_substream *substream = siu_stream->substream; struct device *dev = substream->pcm->card->dev; struct snd_pcm_runtime *rt = substream->runtime; @@ -520,10 +520,8 @@ static int siu_pcm_new(struct snd_soc_component *component, (*port_info)->pcm = pcm; /* IO tasklets */ - tasklet_init(&(*port_info)->playback.tasklet, siu_io_tasklet, - (unsigned long)&(*port_info)->playback); - tasklet_init(&(*port_info)->capture.tasklet, siu_io_tasklet, - (unsigned long)&(*port_info)->capture); + tasklet_setup(&(*port_info)->playback.tasklet, siu_io_tasklet); + tasklet_setup(&(*port_info)->capture.tasklet, siu_io_tasklet); } dev_info(card->dev, "SuperH SIU driver initialized.\n"); diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index f0b4f4bc44a4..5504b92946e3 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -406,7 +406,7 @@ static unsigned int soc_component_read_no_lock( ret = -EIO; if (ret < 0) - soc_component_ret(component, ret); + return soc_component_ret(component, ret); return val; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2fe1b2ec7c8f..054437660678 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -618,7 +618,7 @@ int snd_soc_suspend(struct device *dev) "ASoC: idle_bias_off CODEC on over suspend\n"); break; } - /* fall through */ + fallthrough; case SND_SOC_BIAS_OFF: snd_soc_component_suspend(component); @@ -834,6 +834,19 @@ struct snd_soc_dai *snd_soc_find_dai( } EXPORT_SYMBOL_GPL(snd_soc_find_dai); +struct snd_soc_dai *snd_soc_find_dai_with_mutex( + const struct snd_soc_dai_link_component *dlc) +{ + struct snd_soc_dai *dai; + + mutex_lock(&client_mutex); + dai = snd_soc_find_dai(dlc); + mutex_unlock(&client_mutex); + + return dai; +} +EXPORT_SYMBOL_GPL(snd_soc_find_dai_with_mutex); + static int soc_dai_link_sanity_check(struct snd_soc_card *card, struct snd_soc_dai_link *link) { diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 91a2551e4cef..0dbd312aad08 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -412,14 +412,14 @@ void snd_soc_dai_link_set_capabilities(struct snd_soc_dai_link *dai_link) supported_codec = false; for_each_link_cpus(dai_link, i, cpu) { - dai = snd_soc_find_dai(cpu); + dai = snd_soc_find_dai_with_mutex(cpu); if (dai && snd_soc_dai_stream_valid(dai, direction)) { supported_cpu = true; break; } } for_each_link_codecs(dai_link, i, codec) { - dai = snd_soc_find_dai(codec); + dai = snd_soc_find_dai_with_mutex(codec); if (dai && snd_soc_dai_stream_valid(dai, direction)) { supported_codec = true; break; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 00ac1cbf6f88..4c9d4cd8cf0b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -812,7 +812,7 @@ dynamic: return 0; config_err: - for_each_rtd_dais(rtd, i, dai) + for_each_rtd_dais_rollback(rtd, i, dai) snd_soc_dai_shutdown(dai, substream); snd_soc_link_shutdown(substream); diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index cee998671318..5b60379237bf 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1057,7 +1057,7 @@ static int soc_tplg_denum_create(struct soc_tplg *tplg, unsigned int count, ec->hdr.name); goto err_denum; } - /* fall through */ + fallthrough; case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: @@ -1445,7 +1445,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( ec->hdr.name); goto err_se; } - /* fall through */ + fallthrough; case SND_SOC_TPLG_CTL_ENUM: case SND_SOC_TPLG_DAPM_CTL_ENUM_DOUBLE: case SND_SOC_TPLG_DAPM_CTL_ENUM_VIRT: diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index df1c6997cb4e..c6cb8c212eca 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -310,7 +310,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream, return ret; } - /* fallthrough */ + fallthrough; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: snd_hdac_ext_link_stream_start(link_dev); @@ -333,7 +333,7 @@ static int hda_link_pcm_trigger(struct snd_pcm_substream *substream, link_dev->link_prepared = 0; - /* fallthrough */ + fallthrough; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: snd_hdac_ext_link_stream_clear(link_dev); break; diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index d730e437e4ba..71c3f29057a7 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -361,7 +361,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component, return ret; } - /* fallthrough */ + fallthrough; case SNDRV_PCM_TRIGGER_START: if (spcm->stream[substream->stream].suspend_ignored) { /* @@ -386,7 +386,7 @@ static int sof_pcm_trigger(struct snd_soc_component *component, spcm->stream[substream->stream].suspend_ignored = true; return 0; } - /* fallthrough */ + fallthrough; case SNDRV_PCM_TRIGGER_STOP: stream.hdr.cmd |= SOF_IPC_STREAM_TRIG_STOP; ipc_first = true; diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index fe7117171a0e..0cbe31e2c7e9 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -71,7 +71,7 @@ static int tegra186_dspk_put_control(struct snd_kcontrol *kcontrol, return 0; } -static int tegra186_dspk_runtime_suspend(struct device *dev) +static int __maybe_unused tegra186_dspk_runtime_suspend(struct device *dev) { struct tegra186_dspk *dspk = dev_get_drvdata(dev); @@ -83,7 +83,7 @@ static int tegra186_dspk_runtime_suspend(struct device *dev) return 0; } -static int tegra186_dspk_runtime_resume(struct device *dev) +static int __maybe_unused tegra186_dspk_runtime_resume(struct device *dev) { struct tegra186_dspk *dspk = dev_get_drvdata(dev); int err; diff --git a/sound/soc/tegra/tegra210_admaif.c b/sound/soc/tegra/tegra210_admaif.c index 4894e8e6ee7f..1268046b345d 100644 --- a/sound/soc/tegra/tegra210_admaif.c +++ b/sound/soc/tegra/tegra210_admaif.c @@ -219,7 +219,7 @@ static const struct regmap_config tegra186_admaif_regmap_config = { .cache_type = REGCACHE_FLAT, }; -static int tegra_admaif_runtime_suspend(struct device *dev) +static int __maybe_unused tegra_admaif_runtime_suspend(struct device *dev) { struct tegra_admaif *admaif = dev_get_drvdata(dev); @@ -229,7 +229,7 @@ static int tegra_admaif_runtime_suspend(struct device *dev) return 0; } -static int tegra_admaif_runtime_resume(struct device *dev) +static int __maybe_unused tegra_admaif_runtime_resume(struct device *dev) { struct tegra_admaif *admaif = dev_get_drvdata(dev); diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c index 5123a96fdde8..66287a7c9865 100644 --- a/sound/soc/tegra/tegra210_ahub.c +++ b/sound/soc/tegra/tegra210_ahub.c @@ -564,7 +564,7 @@ static const struct of_device_id tegra_ahub_of_match[] = { }; MODULE_DEVICE_TABLE(of, tegra_ahub_of_match); -static int tegra_ahub_runtime_suspend(struct device *dev) +static int __maybe_unused tegra_ahub_runtime_suspend(struct device *dev) { struct tegra_ahub *ahub = dev_get_drvdata(dev); @@ -576,7 +576,7 @@ static int tegra_ahub_runtime_suspend(struct device *dev) return 0; } -static int tegra_ahub_runtime_resume(struct device *dev) +static int __maybe_unused tegra_ahub_runtime_resume(struct device *dev) { struct tegra_ahub *ahub = dev_get_drvdata(dev); int err; diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c index d682414ad90d..a661f40bc41c 100644 --- a/sound/soc/tegra/tegra210_dmic.c +++ b/sound/soc/tegra/tegra210_dmic.c @@ -40,7 +40,7 @@ static const struct reg_default tegra210_dmic_reg_defaults[] = { { TEGRA210_DMIC_LP_BIQUAD_1_COEF_4, 0x0 }, }; -static int tegra210_dmic_runtime_suspend(struct device *dev) +static int __maybe_unused tegra210_dmic_runtime_suspend(struct device *dev) { struct tegra210_dmic *dmic = dev_get_drvdata(dev); @@ -52,7 +52,7 @@ static int tegra210_dmic_runtime_suspend(struct device *dev) return 0; } -static int tegra210_dmic_runtime_resume(struct device *dev) +static int __maybe_unused tegra210_dmic_runtime_resume(struct device *dev) { struct tegra210_dmic *dmic = dev_get_drvdata(dev); int err; diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index 722092181583..a383bd5c51cd 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -164,7 +164,7 @@ static int tegra210_i2s_init(struct snd_soc_dapm_widget *w, return tegra210_i2s_sw_reset(compnt, is_playback); } -static int tegra210_i2s_runtime_suspend(struct device *dev) +static int __maybe_unused tegra210_i2s_runtime_suspend(struct device *dev) { struct tegra210_i2s *i2s = dev_get_drvdata(dev); @@ -176,7 +176,7 @@ static int tegra210_i2s_runtime_suspend(struct device *dev) return 0; } -static int tegra210_i2s_runtime_resume(struct device *dev) +static int __maybe_unused tegra210_i2s_runtime_resume(struct device *dev) { struct tegra210_i2s *i2s = dev_get_drvdata(dev); int err; diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 5c47de96c529..57feb473a579 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -446,12 +446,12 @@ static const struct snd_soc_dai_ops ams_delta_dai_ops = { /* Will be used if the codec ever has its own digital_mute function */ static int ams_delta_startup(struct snd_pcm_substream *substream) { - return ams_delta_digital_mute(NULL, 0, substream->stream); + return ams_delta_mute(NULL, 0, substream->stream); } static void ams_delta_shutdown(struct snd_pcm_substream *substream) { - ams_delta_digital_mute(NULL, 1, substream->stream); + ams_delta_mute(NULL, 1, substream->stream); } diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index d89b5c928c4d..dd34504c09ba 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -289,7 +289,7 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, * rate is lowered. */ inv_fs = true; - /* fall through */ + fallthrough; case SND_SOC_DAIFMT_DSP_A: dev->mode = MOD_DSP_A; break; diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c index 2802a33b9c5f..ed217b34f846 100644 --- a/sound/soc/ti/n810.c +++ b/sound/soc/ti/n810.c @@ -46,7 +46,7 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm) switch (n810_jack_func) { case N810_JACK_HS: line1l = 1; - /* fall through */ + fallthrough; case N810_JACK_HP: hp = 1; break; diff --git a/sound/soc/ti/omap-dmic.c b/sound/soc/ti/omap-dmic.c index 01abf1be5d78..a26588e9c3bc 100644 --- a/sound/soc/ti/omap-dmic.c +++ b/sound/soc/ti/omap-dmic.c @@ -203,10 +203,10 @@ static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream, switch (channels) { case 6: dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE; - /* fall through */ + fallthrough; case 4: dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE; - /* fall through */ + fallthrough; case 2: dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE; break; diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c index d482b62f314a..fafb2998ad0d 100644 --- a/sound/soc/ti/omap-mcpdm.c +++ b/sound/soc/ti/omap-mcpdm.c @@ -309,19 +309,19 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 4; - /* fall through */ + fallthrough; case 4: if (stream == SNDRV_PCM_STREAM_CAPTURE) /* up to 3 channels for capture */ return -EINVAL; link_mask |= 1 << 3; - /* fall through */ + fallthrough; case 3: link_mask |= 1 << 2; - /* fall through */ + fallthrough; case 2: link_mask |= 1 << 1; - /* fall through */ + fallthrough; case 1: link_mask |= 1 << 0; break; diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c index 2176a95201bf..a2629ccc1dc8 100644 --- a/sound/soc/ti/rx51.c +++ b/sound/soc/ti/rx51.c @@ -55,7 +55,7 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm) break; case RX51_JACK_HS: hs = 1; - /* fall through */ + fallthrough; case RX51_JACK_HP: hp = 1; break; diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 4b1cd4da3e36..939b33ec39f5 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -134,9 +134,9 @@ txx9aclc_dma_submit(struct txx9aclc_dmadata *dmadata, dma_addr_t buf_dma_addr) #define NR_DMA_CHAIN 2 -static void txx9aclc_dma_tasklet(unsigned long data) +static void txx9aclc_dma_tasklet(struct tasklet_struct *t) { - struct txx9aclc_dmadata *dmadata = (struct txx9aclc_dmadata *)data; + struct txx9aclc_dmadata *dmadata = from_tasklet(dmadata, t, tasklet); struct dma_chan *chan = dmadata->dma_chan; struct dma_async_tx_descriptor *desc; struct snd_pcm_substream *substream = dmadata->substream; @@ -352,8 +352,7 @@ static int txx9aclc_dma_init(struct txx9aclc_soc_device *dev, "playback" : "capture"); return -EBUSY; } - tasklet_init(&dmadata->tasklet, txx9aclc_dma_tasklet, - (unsigned long)dmadata); + tasklet_setup(&dmadata->tasklet, txx9aclc_dma_tasklet); return 0; } diff --git a/sound/soc/zte/zx-i2s.c b/sound/soc/zte/zx-i2s.c index 568cde64ff8b..1c1a44e08a67 100644 --- a/sound/soc/zte/zx-i2s.c +++ b/sound/soc/zte/zx-i2s.c @@ -294,7 +294,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, zx_i2s_rx_dma_en(zx_i2s->reg_base, true); else zx_i2s_tx_dma_en(zx_i2s->reg_base, true); - /* fall thru */ + fallthrough; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (capture) @@ -308,7 +308,7 @@ static int zx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, zx_i2s_rx_dma_en(zx_i2s->reg_base, false); else zx_i2s_tx_dma_en(zx_i2s->reg_base, false); - /* fall thru */ + fallthrough; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: if (capture) diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c index a3a07c0730e6..b4168bd532b7 100644 --- a/sound/soc/zte/zx-spdif.c +++ b/sound/soc/zte/zx-spdif.c @@ -218,7 +218,7 @@ static int zx_spdif_trigger(struct snd_pcm_substream *substream, int cmd, val = readl_relaxed(zx_spdif->reg_base + ZX_FIFOCTRL); val |= ZX_FIFOCTRL_TX_FIFO_RST; writel_relaxed(val, zx_spdif->reg_base + ZX_FIFOCTRL); - /* fall thru */ + fallthrough; case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: zx_spdif_cfg_tx(zx_spdif->reg_base, true); diff --git a/sound/usb/midi.c b/sound/usb/midi.c index df639fe03118..e8287a05e36b 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -344,10 +344,9 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep) spin_unlock_irqrestore(&ep->buffer_lock, flags); } -static void snd_usbmidi_out_tasklet(unsigned long data) +static void snd_usbmidi_out_tasklet(struct tasklet_struct *t) { - struct snd_usb_midi_out_endpoint *ep = - (struct snd_usb_midi_out_endpoint *) data; + struct snd_usb_midi_out_endpoint *ep = from_tasklet(ep, t, tasklet); snd_usbmidi_do_output(ep); } @@ -1441,7 +1440,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi, } spin_lock_init(&ep->buffer_lock); - tasklet_init(&ep->tasklet, snd_usbmidi_out_tasklet, (unsigned long)ep); + tasklet_setup(&ep->tasklet, snd_usbmidi_out_tasklet); init_waitqueue_head(&ep->drain_wait); for (i = 0; i < 0x10; ++i) diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 884e740a785c..3b2dce1043f5 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -247,9 +247,9 @@ static inline void add_with_wraparound(struct ua101 *ua, *value -= ua->playback.queue_length; } -static void playback_tasklet(unsigned long data) +static void playback_tasklet(struct tasklet_struct *t) { - struct ua101 *ua = (void *)data; + struct ua101 *ua = from_tasklet(ua, t, playback_tasklet); unsigned long flags; unsigned int frames; struct ua101_urb *urb; @@ -1218,8 +1218,7 @@ static int ua101_probe(struct usb_interface *interface, spin_lock_init(&ua->lock); mutex_init(&ua->mutex); INIT_LIST_HEAD(&ua->ready_playback_urbs); - tasklet_init(&ua->playback_tasklet, - playback_tasklet, (unsigned long)ua); + tasklet_setup(&ua->playback_tasklet, playback_tasklet); init_waitqueue_head(&ua->alsa_capture_wait); init_waitqueue_head(&ua->rate_feedback_wait); init_waitqueue_head(&ua->alsa_playback_wait); diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6b0f3a8469ef..81e987eaf063 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2371,7 +2371,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, int num_ins; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; - int i, err, nameid, type, len; + int i, err, nameid, type, len, val; const struct procunit_info *info; const struct procunit_value_info *valinfo; const struct usbmix_name_map *map; @@ -2474,6 +2474,12 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, break; } + err = get_cur_ctl_value(cval, cval->control << 8, &val); + if (err < 0) { + usb_mixer_elem_info_free(cval); + return -EINVAL; + } + kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (!kctl) { usb_mixer_elem_info_free(cval); diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 5b43e9e40e49..c369c81e74c4 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -371,7 +371,6 @@ static const struct usbmix_name_map asus_rog_map[] = { }; static const struct usbmix_name_map lenovo_p620_rear_map[] = { - { 19, NULL, 2 }, /* FU, Volume */ { 19, NULL, 12 }, /* FU, Input Gain Pad */ {} }; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5600751803cf..b401ee894e1b 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -369,11 +369,13 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, case USB_ID(0x07fd, 0x0008): /* MOTU M Series */ case USB_ID(0x31e9, 0x0001): /* Solid State Logic SSL2 */ case USB_ID(0x31e9, 0x0002): /* Solid State Logic SSL2+ */ + case USB_ID(0x0499, 0x172f): /* Steinberg UR22C */ case USB_ID(0x0d9a, 0x00df): /* RTX6001 */ ep = 0x81; ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ ep = 0x82; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index d79e3ddc5690..23eafd50126f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2680,6 +2680,10 @@ YAMAHA_DEVICE(0x7010, "UB99"), .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, @@ -2690,6 +2694,32 @@ YAMAHA_DEVICE(0x7010, "UB99"), .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x01, .ep_attr = USB_ENDPOINT_XFER_ISOC, + .datainterval = 1, + .maxpacksize = 0x024c, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .datainterval = 1, + .maxpacksize = 0x0126, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .rate_min = 44100, @@ -2797,14 +2827,24 @@ YAMAHA_DEVICE(0x7010, "UB99"), /* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */ { USB_DEVICE(0x17aa, 0x1046), - QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Rear", - "Lenovo-ThinkStation-P620-Rear"), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Lenovo", + .product_name = "ThinkStation P620 Rear", + .profile_name = "Lenovo-ThinkStation-P620-Rear", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND + } }, /* Lenovo ThinkStation P620 Internal Speaker + Front Headset */ { USB_DEVICE(0x17aa, 0x104d), - QUIRK_DEVICE_PROFILE("Lenovo", "ThinkStation P620 Main", - "Lenovo-ThinkStation-P620-Main"), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Lenovo", + .product_name = "ThinkStation P620 Main", + .profile_name = "Lenovo-ThinkStation-P620-Main", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_SETUP_DISABLE_AUTOSUSPEND + } }, /* Native Instruments MK2 series */ @@ -3519,14 +3559,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), { /* * Pioneer DJ DJM-250MK2 - * PCM is 8 channels out @ 48 fixed (endpoints 0x01). - * The output from computer to the mixer is usable. + * PCM is 8 channels out @ 48 fixed (endpoint 0x01) + * and 8 channels in @ 48 fixed (endpoint 0x82). + * + * Both playback and recording is working, even simultaneously. + * + * Playback channels could be mapped to: + * - CH1 + * - CH2 + * - AUX * - * The input (phono or line to computer) is not working. - * It should be at endpoint 0x82 and probably also 8 channels, - * but it seems that it works only with Pioneer proprietary software. - * Even on officially supported OS, the Audacity was unable to record - * and Mixxx to recognize the control vinyls. + * Recording channels could be mapped to: + * - Post CH1 Fader + * - Post CH2 Fader + * - Cross Fader A + * - Cross Fader B + * - MIC + * - AUX + * - REC OUT + * + * There is remaining problem with recording directly from PHONO/LINE. + * If we map a channel to: + * - CH1 Control Tone PHONO + * - CH1 Control Tone LINE + * - CH2 Control Tone PHONO + * - CH2 Control Tone LINE + * it is silent. + * There is no signal even on other operating systems with official drivers. + * The signal appears only when a supported application is started. + * This needs to be investigated yet... + * (there is quite a lot communication on the USB in both directions) + * + * In current version this mixer could be used for playback + * and for recording from vinyls (through Post CH* Fader) + * but not for DVS (Digital Vinyl Systems) like in Mixxx. */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { @@ -3550,6 +3616,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .rate_max = 48000, .nr_rates = 1, .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 8, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } } }, { @@ -3714,8 +3800,8 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */ * they pretend to be 96kHz mono as a workaround for stereo being broken * by that... * - * They also have swapped L-R channels, but that's for userspace to deal - * with. + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. */ { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index abf99b814a0f..892296df131d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -518,6 +518,15 @@ static int setup_fmt_after_resume_quirk(struct snd_usb_audio *chip, return 1; /* Continue with creating streams and mixer */ } +static int setup_disable_autosuspend(struct snd_usb_audio *chip, + struct usb_interface *iface, + struct usb_driver *driver, + const struct snd_usb_audio_quirk *quirk) +{ + driver->supports_autosuspend = 0; + return 1; /* Continue with creating streams and mixer */ +} + /* * audio-interface quirks * @@ -557,6 +566,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk, [QUIRK_AUDIO_STANDARD_MIXER] = create_standard_mixer_quirk, [QUIRK_SETUP_FMT_AFTER_RESUME] = setup_fmt_after_resume_quirk, + [QUIRK_SETUP_DISABLE_AUTOSUSPEND] = setup_disable_autosuspend, }; if (quirk->type < QUIRK_TYPE_COUNT) { @@ -1493,6 +1503,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, set_format_emu_quirk(subs, fmt); break; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ pioneer_djm_set_format_quirk(subs); break; case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ @@ -1667,12 +1678,13 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); - /* Zoom R16/24, Logitech H650e, Jabra 550a, Kingston HyperX needs a tiny - * delay here, otherwise requests like get/set frequency return as - * failed despite actually succeeding. + /* Zoom R16/24, Logitech H650e/H570e, Jabra 550a, Kingston HyperX + * needs a tiny delay here, otherwise requests like get/set + * frequency return as failed despite actually succeeding. */ if ((chip->usb_id == USB_ID(0x1686, 0x00dd) || chip->usb_id == USB_ID(0x046d, 0x0a46) || + chip->usb_id == USB_ID(0x046d, 0x0a56) || chip->usb_id == USB_ID(0x0b0e, 0x0349) || chip->usb_id == USB_ID(0x0951, 0x16ad)) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index b91c4c0807ec..6839915a0128 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -102,6 +102,7 @@ enum quirk_type { QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_AUDIO_STANDARD_MIXER, QUIRK_SETUP_FMT_AFTER_RESUME, + QUIRK_SETUP_DISABLE_AUTOSUSPEND, QUIRK_TYPE_COUNT }; diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig index 77777192f650..4ffcc5e623c2 100644 --- a/sound/x86/Kconfig +++ b/sound/x86/Kconfig @@ -9,7 +9,7 @@ menuconfig SND_X86 if SND_X86 config HDMI_LPE_AUDIO - tristate "HDMI audio without HDaudio on Intel Atom platforms" + tristate "HDMI audio without HDAudio on Intel Atom platforms" depends on DRM_I915 select SND_PCM help |