diff options
38 files changed, 281 insertions, 105 deletions
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 8b6aeb8a78f7..02fd65993e7e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2155,6 +2155,8 @@ int pcm_lib_apply_appl_ptr(struct snd_pcm_substream *substream, ret = substream->ops->ack(substream); if (ret < 0) { runtime->control->appl_ptr = old_appl_ptr; + if (ret == -EPIPE) + __snd_pcm_xrun(substream); return ret; } } diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index ae31bb127594..317bdf6dcbef 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -472,6 +472,15 @@ static const struct config_entry config_table[] = { }, #endif +/* Meteor Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_METEORLAKE) + /* Meteorlake-P */ + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x7e28, + }, +#endif + }; static const struct config_entry *snd_intel_dsp_find_config diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 27e11b5f70b9..c7d7eff86727 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -430,7 +430,7 @@ void HPI_6205(struct hpi_message *phm, struct hpi_response *phr) pao = hpi_find_adapter(phm->adapter_index); } else { /* subsys messages don't address an adapter */ - _HPI_6205(NULL, phm, phr); + phr->error = HPI_ERROR_INVALID_OBJ_INDEX; return; } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 81c4a45254ff..77a592f21947 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -328,14 +328,15 @@ enum { #define needs_eld_notify_link(chip) false #endif -#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \ +#define CONTROLLER_IN_GPU(pci) (((pci)->vendor == 0x8086) && \ + (((pci)->device == 0x0a0c) || \ ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ ((pci)->device == 0x490d) || \ ((pci)->device == 0x4f90) || \ ((pci)->device == 0x4f91) || \ - ((pci)->device == 0x4f92)) + ((pci)->device == 0x4f92))) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index acde4cd58785..099722ebaed8 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -4228,8 +4228,10 @@ static int tuning_ctl_set(struct hda_codec *codec, hda_nid_t nid, for (i = 0; i < TUNING_CTLS_COUNT; i++) if (nid == ca0132_tuning_ctls[i].nid) - break; + goto found; + return -EINVAL; +found: snd_hda_power_up(codec); dspio_set_param(codec, ca0132_tuning_ctls[i].mid, 0x20, ca0132_tuning_ctls[i].req, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 75e1d00074b9..a889cccdd607 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -980,7 +980,10 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_PINCFG_LENOVO_NOTEBOOK), + /* NOTE: we'd need to extend the quirk for 17aa:3977 as the same + * PCI SSID is used on multiple Lenovo models + */ + SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), @@ -1003,6 +1006,7 @@ static const struct hda_model_fixup cxt5066_fixup_models[] = { { .id = CXT_FIXUP_MUTE_LED_GPIO, .name = "mute-led-gpio" }, { .id = CXT_FIXUP_HP_ZBOOK_MUTE_LED, .name = "hp-zbook-mute-led" }, { .id = CXT_FIXUP_HP_MIC_NO_PRESENCE, .name = "hp-mic-fix" }, + { .id = CXT_PINCFG_LENOVO_NOTEBOOK, .name = "lenovo-20149" }, {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3c629f4ae080..b501f9489fc1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2631,6 +2631,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x65e5, "Clevo PC50D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f5, "Clevo PD50PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x66a2, "Clevo PE60RNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -2651,6 +2652,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x96e1, "Clevo P960[ER][CDFN]-K", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x97e1, "Clevo P970[ER][CDFN]", ALC1220_FIXUP_CLEVO_P950), SND_PCI_QUIRK(0x1558, 0x97e2, "Clevo P970RC-M", ALC1220_FIXUP_CLEVO_P950), + SND_PCI_QUIRK(0x1558, 0xd502, "Clevo PD50SNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_FIXUP_EAPD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", ALC882_FIXUP_LENOVO_Y530), @@ -9260,7 +9262,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC295_FIXUP_CHROME_BOOK), + SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), @@ -9447,6 +9449,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8b8a, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8b, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b8d, "HP", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8b8f, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8b92, "HP", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8bf0, "HP", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), @@ -9539,6 +9542,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), + SND_PCI_QUIRK(0x144d, 0xc868, "Samsung Galaxy Book2 Pro (NP930XED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), @@ -9573,6 +9577,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x5101, "Clevo S510WU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x5157, "Clevo W517GU1", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x51a1, "Clevo NS50MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x5630, "Clevo NP50RNJS", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70a1, "Clevo NB70T[HJK]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70b3, "Clevo NK70SB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f2, "Clevo NH79EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), @@ -9607,6 +9612,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb022, "Clevo NH77D[DC][QW]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 4a4f09f924bc..e3d398b8f54e 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -968,6 +968,8 @@ int da7219_aad_init(struct snd_soc_component *component) INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work); INIT_WORK(&da7219_aad->jack_det_work, da7219_aad_jack_det_work); + mutex_init(&da7219_aad->jack_det_mutex); + ret = request_threaded_irq(da7219_aad->irq, da7219_aad_pre_irq_thread, da7219_aad_irq_thread, IRQF_TRIGGER_LOW | IRQF_ONESHOT, diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 01e8ffda2a4b..6d980fbc4207 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -428,8 +428,13 @@ static int hdmi_codec_startup(struct snd_pcm_substream *substream, { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool has_capture = !hcp->hcd.no_i2s_capture; + bool has_playback = !hcp->hcd.no_i2s_playback; int ret = 0; + if (!((has_playback && tx) || (has_capture && !tx))) + return 0; + mutex_lock(&hcp->lock); if (hcp->busy) { dev_err(dai->dev, "Only one simultaneous stream supported!\n"); @@ -468,6 +473,12 @@ static void hdmi_codec_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool has_capture = !hcp->hcd.no_i2s_capture; + bool has_playback = !hcp->hcd.no_i2s_playback; + + if (!((has_playback && tx) || (has_capture && !tx))) + return; hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; hcp->hcd.ops->audio_shutdown(dai->dev->parent, hcp->hcd.data); diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index bf27bdd5be20..473d3cd39554 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -242,7 +242,7 @@ enum { struct tx_mute_work { struct tx_macro *tx; - u32 decimator; + u8 decimator; struct delayed_work dwork; }; @@ -635,7 +635,7 @@ exit: return 0; } -static bool is_amic_enabled(struct snd_soc_component *component, int decimator) +static bool is_amic_enabled(struct snd_soc_component *component, u8 decimator) { u16 adc_mux_reg, adc_reg, adc_n; @@ -849,7 +849,7 @@ static int tx_macro_enable_dec(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); - unsigned int decimator; + u8 decimator; u16 tx_vol_ctl_reg, dec_cfg_reg, hpf_gate_reg, tx_gain_ctl_reg; u8 hpf_cut_off_freq; int hpf_delay = TX_MACRO_DMIC_HPF_DELAY_MS; @@ -1064,7 +1064,8 @@ static int tx_macro_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; - u32 decimator, sample_rate; + u32 sample_rate; + u8 decimator; int tx_fs_rate; struct tx_macro *tx = snd_soc_component_get_drvdata(component); @@ -1128,7 +1129,7 @@ static int tx_macro_digital_mute(struct snd_soc_dai *dai, int mute, int stream) { struct snd_soc_component *component = dai->component; struct tx_macro *tx = snd_soc_component_get_drvdata(component); - u16 decimator; + u8 decimator; /* active decimator not set yet */ if (tx->active_decimator[dai->id] == -1) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 614eceda6b9e..33b67db8794e 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -294,6 +294,10 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. + Note that this is an old driver. Consider enabling + SND_SOC_FSL_ASOC_CARD and SND_SOC_SGTL5000 to use the newer + driver. + config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index acd43b6108e9..1a1d572cc1d0 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -117,6 +117,26 @@ static void avs_da7219_codec_exit(struct snd_soc_pcm_runtime *rtd) snd_soc_component_set_jack(asoc_rtd_to_codec(rtd, 0)->component, NULL, NULL); } +static int +avs_da7219_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, struct snd_soc_dai_link **dai_link) { @@ -148,6 +168,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in dl->num_platforms = 1; dl->id = 0; dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->be_hw_params_fixup = avs_da7219_be_fixup; dl->init = avs_da7219_codec_init; dl->exit = avs_da7219_codec_exit; dl->nonatomic = 1; diff --git a/sound/soc/intel/avs/boards/max98357a.c b/sound/soc/intel/avs/boards/max98357a.c index 921f42caf7e0..183123d08c5a 100644 --- a/sound/soc/intel/avs/boards/max98357a.c +++ b/sound/soc/intel/avs/boards/max98357a.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-acpi.h> #include <sound/soc-dapm.h> @@ -24,6 +25,26 @@ static const struct snd_soc_dapm_route card_base_routes[] = { { "Spk", NULL, "Speaker" }, }; +static int +avs_max98357a_be_fixup(struct snd_soc_pcm_runtime *runrime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, struct snd_soc_dai_link **dai_link) { @@ -55,6 +76,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in dl->num_platforms = 1; dl->id = 0; dl->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS; + dl->be_hw_params_fixup = avs_max98357a_be_fixup; dl->nonatomic = 1; dl->no_pcm = 1; dl->dpcm_playback = 1; diff --git a/sound/soc/intel/avs/boards/nau8825.c b/sound/soc/intel/avs/boards/nau8825.c index b31fa931ba8b..b69fc5567135 100644 --- a/sound/soc/intel/avs/boards/nau8825.c +++ b/sound/soc/intel/avs/boards/nau8825.c @@ -33,15 +33,15 @@ avs_nau8825_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *co return -EINVAL; } - if (!SND_SOC_DAPM_EVENT_ON(event)) { + if (SND_SOC_DAPM_EVENT_ON(event)) + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000, + SND_SOC_CLOCK_IN); + else ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return ret; - } - } + if (ret < 0) + dev_err(card->dev, "Set sysclk failed: %d\n", ret); - return 0; + return ret; } static const struct snd_kcontrol_new card_controls[] = { diff --git a/sound/soc/intel/avs/boards/rt5682.c b/sound/soc/intel/avs/boards/rt5682.c index 473e9fe5d0bf..b2c2ba93dcb5 100644 --- a/sound/soc/intel/avs/boards/rt5682.c +++ b/sound/soc/intel/avs/boards/rt5682.c @@ -169,6 +169,27 @@ static const struct snd_soc_ops avs_rt5682_ops = { .hw_params = avs_rt5682_hw_params, }; +static int +avs_rt5682_be_fixup(struct snd_soc_pcm_runtime *runtime, struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate, *channels; + struct snd_mask *fmt; + + rate = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSPN to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + static int avs_create_dai_link(struct device *dev, const char *platform_name, int ssp_port, struct snd_soc_dai_link **dai_link) { @@ -201,6 +222,7 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in dl->id = 0; dl->init = avs_rt5682_codec_init; dl->exit = avs_rt5682_codec_exit; + dl->be_hw_params_fixup = avs_rt5682_be_fixup; dl->ops = &avs_rt5682_ops; dl->nonatomic = 1; dl->no_pcm = 1; diff --git a/sound/soc/intel/avs/boards/ssm4567.c b/sound/soc/intel/avs/boards/ssm4567.c index c5db69612762..2b7f5ad92aca 100644 --- a/sound/soc/intel/avs/boards/ssm4567.c +++ b/sound/soc/intel/avs/boards/ssm4567.c @@ -15,7 +15,6 @@ #include <sound/soc-acpi.h> #include "../../../codecs/nau8825.h" -#define SKL_NUVOTON_CODEC_DAI "nau8825-hifi" #define SKL_SSM_CODEC_DAI "ssm4567-hifi" static struct snd_soc_codec_conf card_codec_conf[] = { @@ -34,41 +33,11 @@ static const struct snd_kcontrol_new card_controls[] = { SOC_DAPM_PIN_SWITCH("Right Speaker"), }; -static int -platform_clock_control(struct snd_soc_dapm_widget *w, struct snd_kcontrol *control, int event) -{ - struct snd_soc_dapm_context *dapm = w->dapm; - struct snd_soc_card *card = dapm->card; - struct snd_soc_dai *codec_dai; - int ret; - - codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); - if (!codec_dai) { - dev_err(card->dev, "Codec dai not found\n"); - return -EINVAL; - } - - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_MCLK, 24000000, - SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(card->dev, "set sysclk err = %d\n", ret); - } else { - ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); - if (ret < 0) - dev_err(card->dev, "set sysclk err = %d\n", ret); - } - - return ret; -} - static const struct snd_soc_dapm_widget card_widgets[] = { SND_SOC_DAPM_SPK("Left Speaker", NULL), SND_SOC_DAPM_SPK("Right Speaker", NULL), SND_SOC_DAPM_SPK("DP1", NULL), SND_SOC_DAPM_SPK("DP2", NULL), - SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, platform_clock_control, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route card_base_routes[] = { diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index 56ee5fef66a8..28dd2046e4ac 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -559,7 +559,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &essx_83x6, .drv_name = "sof-essx8336", - .sof_tplg_filename = "sof-adl-es83x6", /* the tplg suffix is added at run time */ + .sof_tplg_filename = "sof-adl-es8336", /* the tplg suffix is added at run time */ .tplg_quirk_mask = SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER | SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, diff --git a/sound/soc/qcom/qdsp6/q6prm.c b/sound/soc/qcom/qdsp6/q6prm.c index 3aa63aac4a68..81554d202658 100644 --- a/sound/soc/qcom/qdsp6/q6prm.c +++ b/sound/soc/qcom/qdsp6/q6prm.c @@ -184,9 +184,9 @@ int q6prm_set_lpass_clock(struct device *dev, int clk_id, int clk_attr, int clk_ unsigned int freq) { if (freq) - return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); + return q6prm_request_lpass_clock(dev, clk_id, clk_attr, clk_root, freq); - return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_attr, freq); + return q6prm_release_lpass_clock(dev, clk_id, clk_attr, clk_root, freq); } EXPORT_SYMBOL_GPL(q6prm_set_lpass_clock); diff --git a/sound/soc/sof/intel/hda-ctrl.c b/sound/soc/sof/intel/hda-ctrl.c index 3aea36c077c9..f3bdeba28412 100644 --- a/sound/soc/sof/intel/hda-ctrl.c +++ b/sound/soc/sof/intel/hda-ctrl.c @@ -196,12 +196,15 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev) goto err; } + usleep_range(500, 1000); + /* exit HDA controller reset */ ret = hda_dsp_ctrl_link_reset(sdev, false); if (ret < 0) { dev_err(sdev->dev, "error: failed to exit HDA controller reset\n"); goto err; } + usleep_range(1000, 1200); hda_codec_detect_mask(sdev); diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 68eb06f13a1f..a6f2822401e0 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -392,6 +392,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) snd_sof_dsp_update8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset, SOF_HDA_VS_D0I3C_I3, value); + /* + * The value written to the D0I3C::I3 bit may not be taken into account immediately. + * A delay is recommended before checking if D0I3C::CIP is cleared + */ + usleep_range(30, 40); + /* Wait for cmd in progress to be cleared before exiting the function */ ret = hda_dsp_wait_d0i3c_done(sdev); if (ret < 0) { @@ -400,6 +406,12 @@ static int hda_dsp_update_d0i3c_register(struct snd_sof_dev *sdev, u8 value) } reg = snd_sof_dsp_read8(sdev, HDA_DSP_HDA_BAR, chip->d0i3_offset); + /* Confirm d0i3 state changed with paranoia check */ + if ((reg ^ value) & SOF_HDA_VS_D0I3C_I3) { + dev_err(sdev->dev, "failed to update D0I3C!\n"); + return -EIO; + } + trace_sof_intel_D0I3C_updated(sdev, reg); return 0; diff --git a/sound/soc/sof/intel/pci-apl.c b/sound/soc/sof/intel/pci-apl.c index 69279dcc92dc..aff6cb573c27 100644 --- a/sound/soc/sof/intel/pci-apl.c +++ b/sound/soc/sof/intel/pci-apl.c @@ -78,6 +78,7 @@ static const struct sof_dev_desc glk_desc = { .nocodec_tplg_filename = "sof-glk-nocodec.tplg", .ops = &sof_apl_ops, .ops_init = sof_apl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ diff --git a/sound/soc/sof/intel/pci-cnl.c b/sound/soc/sof/intel/pci-cnl.c index 8db3f8d15b55..4c0c1c369dcd 100644 --- a/sound/soc/sof/intel/pci-cnl.c +++ b/sound/soc/sof/intel/pci-cnl.c @@ -48,6 +48,7 @@ static const struct sof_dev_desc cnl_desc = { .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .ops_init = sof_cnl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc cfl_desc = { @@ -111,6 +112,7 @@ static const struct sof_dev_desc cml_desc = { .nocodec_tplg_filename = "sof-cnl-nocodec.tplg", .ops = &sof_cnl_ops, .ops_init = sof_cnl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ diff --git a/sound/soc/sof/intel/pci-icl.c b/sound/soc/sof/intel/pci-icl.c index d6cf75e357db..6785669113b3 100644 --- a/sound/soc/sof/intel/pci-icl.c +++ b/sound/soc/sof/intel/pci-icl.c @@ -79,6 +79,7 @@ static const struct sof_dev_desc jsl_desc = { .nocodec_tplg_filename = "sof-jsl-nocodec.tplg", .ops = &sof_cnl_ops, .ops_init = sof_cnl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ diff --git a/sound/soc/sof/intel/pci-mtl.c b/sound/soc/sof/intel/pci-mtl.c index 6e4e6d4ef5a5..b183dc0014b4 100644 --- a/sound/soc/sof/intel/pci-mtl.c +++ b/sound/soc/sof/intel/pci-mtl.c @@ -46,6 +46,7 @@ static const struct sof_dev_desc mtl_desc = { .nocodec_tplg_filename = "sof-mtl-nocodec.tplg", .ops = &sof_mtl_ops, .ops_init = sof_mtl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ diff --git a/sound/soc/sof/intel/pci-skl.c b/sound/soc/sof/intel/pci-skl.c index 3a99dc444f92..5b4bccf81965 100644 --- a/sound/soc/sof/intel/pci-skl.c +++ b/sound/soc/sof/intel/pci-skl.c @@ -38,6 +38,7 @@ static struct sof_dev_desc skl_desc = { .nocodec_tplg_filename = "sof-skl-nocodec.tplg", .ops = &sof_skl_ops, .ops_init = sof_skl_ops_init, + .ops_free = hda_ops_free, }; static struct sof_dev_desc kbl_desc = { @@ -61,6 +62,7 @@ static struct sof_dev_desc kbl_desc = { .nocodec_tplg_filename = "sof-kbl-nocodec.tplg", .ops = &sof_skl_ops, .ops_init = sof_skl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ diff --git a/sound/soc/sof/intel/pci-tgl.c b/sound/soc/sof/intel/pci-tgl.c index e80c4dfef85a..22e769e0831d 100644 --- a/sound/soc/sof/intel/pci-tgl.c +++ b/sound/soc/sof/intel/pci-tgl.c @@ -48,6 +48,7 @@ static const struct sof_dev_desc tgl_desc = { .nocodec_tplg_filename = "sof-tgl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc tglh_desc = { @@ -110,6 +111,7 @@ static const struct sof_dev_desc ehl_desc = { .nocodec_tplg_filename = "sof-ehl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc adls_desc = { @@ -141,6 +143,7 @@ static const struct sof_dev_desc adls_desc = { .nocodec_tplg_filename = "sof-adl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc adl_desc = { @@ -172,6 +175,7 @@ static const struct sof_dev_desc adl_desc = { .nocodec_tplg_filename = "sof-adl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc adl_n_desc = { @@ -203,6 +207,7 @@ static const struct sof_dev_desc adl_n_desc = { .nocodec_tplg_filename = "sof-adl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc rpls_desc = { @@ -234,6 +239,7 @@ static const struct sof_dev_desc rpls_desc = { .nocodec_tplg_filename = "sof-rpl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; static const struct sof_dev_desc rpl_desc = { @@ -265,6 +271,7 @@ static const struct sof_dev_desc rpl_desc = { .nocodec_tplg_filename = "sof-rpl-nocodec.tplg", .ops = &sof_tgl_ops, .ops_init = sof_tgl_ops_init, + .ops_free = hda_ops_free, }; /* PCI IDs */ diff --git a/sound/soc/sof/intel/pci-tng.c b/sound/soc/sof/intel/pci-tng.c index 5b2b409752c5..8c22a00266c0 100644 --- a/sound/soc/sof/intel/pci-tng.c +++ b/sound/soc/sof/intel/pci-tng.c @@ -75,11 +75,7 @@ static int tangier_pci_probe(struct snd_sof_dev *sdev) /* LPE base */ base = pci_resource_start(pci, desc->resindex_lpe_base) - IRAM_OFFSET; - size = pci_resource_len(pci, desc->resindex_lpe_base); - if (size < PCI_BAR_SIZE) { - dev_err(sdev->dev, "error: I/O region is too small.\n"); - return -ENODEV; - } + size = PCI_BAR_SIZE; dev_dbg(sdev->dev, "LPE PHY base at 0x%x size 0x%x", base, size); sdev->bar[DSP_BAR] = devm_ioremap(sdev->dev, base, size); diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index dceb78bfe17c..b1f425b39db9 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2081,7 +2081,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * break; case SOF_DAI_INTEL_ALH: if (data) { - config->dai_index = data->dai_index; + /* save the dai_index during hw_params and reuse it for hw_free */ + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) + config->dai_index = data->dai_index; config->alh.stream_id = data->dai_data; } break; @@ -2089,7 +2091,30 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * break; } - config->flags = flags; + /* + * The dai_config op is invoked several times and the flags argument varies as below: + * BE DAI hw_params: When the op is invoked during the BE DAI hw_params, flags contains + * SOF_DAI_CONFIG_FLAGS_HW_PARAMS along with quirks + * FE DAI hw_params: When invoked during FE DAI hw_params after the DAI widget has + * just been set up in the DSP, flags is set to SOF_DAI_CONFIG_FLAGS_HW_PARAMS with no + * quirks + * BE DAI trigger: When invoked during the BE DAI trigger, flags is set to + * SOF_DAI_CONFIG_FLAGS_PAUSE and contains no quirks + * BE DAI hw_free: When invoked during the BE DAI hw_free, flags is set to + * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks + * FE DAI hw_free: When invoked during the FE DAI hw_free, flags is set to + * SOF_DAI_CONFIG_FLAGS_HW_FREE and contains no quirks + * + * The DAI_CONFIG IPC is sent to the DSP, only after the widget is set up during the FE + * DAI hw_params. But since the BE DAI hw_params precedes the FE DAI hw_params, the quirks + * need to be preserved when assigning the flags before sending the IPC. + * For the case of PAUSE/HW_FREE, since there are no quirks, flags can be used as is. + */ + + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) + config->flags |= flags; + else + config->flags = flags; /* only send the IPC if the widget is set up in the DSP */ if (swidget->use_count > 0) { @@ -2097,6 +2122,9 @@ static int sof_ipc3_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * &reply, sizeof(reply)); if (ret < 0) dev_err(sdev->dev, "Failed to set dai config for %s\n", dai->name); + + /* clear the flags once the IPC has been sent even if it fails */ + config->flags = SOF_DAI_CONFIG_FLAGS_NONE; } return ret; diff --git a/sound/soc/sof/ipc3.c b/sound/soc/sof/ipc3.c index 3de64ea2dc9a..4493bbd7faf1 100644 --- a/sound/soc/sof/ipc3.c +++ b/sound/soc/sof/ipc3.c @@ -970,8 +970,9 @@ static void sof_ipc3_rx_msg(struct snd_sof_dev *sdev) return; } - if (hdr.size < sizeof(hdr)) { - dev_err(sdev->dev, "The received message size is invalid\n"); + if (hdr.size < sizeof(hdr) || hdr.size > SOF_IPC_MSG_MAX_SIZE) { + dev_err(sdev->dev, "The received message size is invalid: %u\n", + hdr.size); return; } diff --git a/sound/soc/sof/ipc4-control.c b/sound/soc/sof/ipc4-control.c index 67bd2233fd9a..9a71af1a613a 100644 --- a/sound/soc/sof/ipc4-control.c +++ b/sound/soc/sof/ipc4-control.c @@ -97,7 +97,8 @@ sof_ipc4_set_volume_data(struct snd_sof_dev *sdev, struct snd_sof_widget *swidge } /* set curve type and duration from topology */ - data.curve_duration = gain->data.curve_duration; + data.curve_duration_l = gain->data.curve_duration_l; + data.curve_duration_h = gain->data.curve_duration_h; data.curve_type = gain->data.curve_type; msg->data_ptr = &data; diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 3e27c7a48ebd..a623707c8ffc 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -107,7 +107,7 @@ static const struct sof_topology_token gain_tokens[] = { get_token_u32, offsetof(struct sof_ipc4_gain_data, curve_type)}, {SOF_TKN_GAIN_RAMP_DURATION, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, - offsetof(struct sof_ipc4_gain_data, curve_duration)}, + offsetof(struct sof_ipc4_gain_data, curve_duration_l)}, {SOF_TKN_GAIN_VAL, SND_SOC_TPLG_TUPLE_TYPE_WORD, get_token_u32, offsetof(struct sof_ipc4_gain_data, init_val)}, }; @@ -155,7 +155,7 @@ static void sof_ipc4_dbg_audio_format(struct device *dev, for (i = 0; i < num_format; i++, ptr = (u8 *)ptr + object_size) { fmt = ptr; dev_dbg(dev, - " #%d: %uKHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x)\n", + " #%d: %uHz, %ubit (ch_map %#x ch_cfg %u interleaving_style %u fmt_cfg %#x)\n", i, fmt->sampling_frequency, fmt->bit_depth, fmt->ch_map, fmt->ch_cfg, fmt->interleaving_style, fmt->fmt_cfg); } @@ -692,7 +692,7 @@ static int sof_ipc4_widget_setup_comp_pga(struct snd_sof_widget *swidget) dev_dbg(scomp->dev, "pga widget %s: ramp type: %d, ramp duration %d, initial gain value: %#x, cpc %d\n", - swidget->widget->name, gain->data.curve_type, gain->data.curve_duration, + swidget->widget->name, gain->data.curve_type, gain->data.curve_duration_l, gain->data.init_val, gain->base_config.cpc); ret = sof_ipc4_widget_setup_msg(swidget, &gain->msg); @@ -980,6 +980,7 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) ipc4_copier = dai->private; if (ipc4_copier->dai_type == SOF_DAI_INTEL_ALH) { + struct sof_ipc4_copier_data *copier_data = &ipc4_copier->data; struct sof_ipc4_alh_configuration_blob *blob; unsigned int group_id; @@ -989,6 +990,9 @@ static void sof_ipc4_unprepare_copier_module(struct snd_sof_widget *swidget) ALH_MULTI_GTW_BASE; ida_free(&alh_group_ida, group_id); } + + /* clear the node ID */ + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; } } @@ -1940,8 +1944,15 @@ static int sof_ipc4_dai_config(struct snd_sof_dev *sdev, struct snd_sof_widget * pipeline->skip_during_fe_trigger = true; fallthrough; case SOF_DAI_INTEL_ALH: - copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; - copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data); + /* + * Do not clear the node ID when this op is invoked with + * SOF_DAI_CONFIG_FLAGS_HW_FREE. It is needed to free the group_ida during + * unprepare. + */ + if (flags & SOF_DAI_CONFIG_FLAGS_HW_PARAMS) { + copier_data->gtw_cfg.node_id &= ~SOF_IPC4_NODE_INDEX_MASK; + copier_data->gtw_cfg.node_id |= SOF_IPC4_NODE_INDEX(data->dai_data); + } break; case SOF_DAI_INTEL_DMIC: case SOF_DAI_INTEL_SSP: diff --git a/sound/soc/sof/ipc4-topology.h b/sound/soc/sof/ipc4-topology.h index 72529179ac22..123f1096f326 100644 --- a/sound/soc/sof/ipc4-topology.h +++ b/sound/soc/sof/ipc4-topology.h @@ -46,7 +46,7 @@ #define SOF_IPC4_NODE_INDEX_INTEL_SSP(x) (((x) & 0xf) << 4) /* Node ID for DMIC type DAI copiers */ -#define SOF_IPC4_NODE_INDEX_INTEL_DMIC(x) (((x) & 0x7) << 5) +#define SOF_IPC4_NODE_INDEX_INTEL_DMIC(x) ((x) & 0x7) #define SOF_IPC4_GAIN_ALL_CHANNELS_MASK 0xffffffff #define SOF_IPC4_VOL_ZERO_DB 0x7fffffff @@ -277,14 +277,16 @@ struct sof_ipc4_control_data { * @init_val: Initial value * @curve_type: Curve type * @reserved: reserved for future use - * @curve_duration: Curve duration + * @curve_duration_l: Curve duration low part + * @curve_duration_h: Curve duration high part */ struct sof_ipc4_gain_data { uint32_t channels; uint32_t init_val; uint32_t curve_type; uint32_t reserved; - uint32_t curve_duration; + uint32_t curve_duration_l; + uint32_t curve_duration_h; } __aligned(8); /** diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 760621bfc802..6de388a8d0b8 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -50,9 +50,27 @@ static int sof_widget_free_unlocked(struct snd_sof_dev *sdev, /* reset route setup status for all routes that contain this widget */ sof_reset_route_setup_status(sdev, swidget); + /* free DAI config and continue to free widget even if it fails */ + if (WIDGET_IS_DAI(swidget->id)) { + struct snd_sof_dai_config_data data; + unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_FREE; + + data.dai_data = DMA_CHAN_INVALID; + + if (tplg_ops && tplg_ops->dai_config) { + err = tplg_ops->dai_config(sdev, swidget, flags, &data); + if (err < 0) + dev_err(sdev->dev, "failed to free config for widget %s\n", + swidget->widget->name); + } + } + /* continue to disable core even if IPC fails */ - if (tplg_ops && tplg_ops->widget_free) - err = tplg_ops->widget_free(sdev, swidget); + if (tplg_ops && tplg_ops->widget_free) { + ret = tplg_ops->widget_free(sdev, swidget); + if (ret < 0 && !err) + err = ret; + } /* * disable widget core. continue to route setup status and complete flag @@ -151,8 +169,12 @@ static int sof_widget_setup_unlocked(struct snd_sof_dev *sdev, /* send config for DAI components */ if (WIDGET_IS_DAI(swidget->id)) { - unsigned int flags = SOF_DAI_CONFIG_FLAGS_NONE; + unsigned int flags = SOF_DAI_CONFIG_FLAGS_HW_PARAMS; + /* + * The config flags saved during BE DAI hw_params will be used for IPC3. IPC4 does + * not use the flags argument. + */ if (tplg_ops && tplg_ops->dai_config) { ret = tplg_ops->dai_config(sdev, swidget, flags, NULL); if (ret < 0) @@ -588,8 +610,8 @@ int sof_widget_list_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params, dir, SOF_WIDGET_SETUP); if (ret < 0) { - ret = sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params, - dir, SOF_WIDGET_UNPREPARE); + sof_walk_widgets_in_order(sdev, spcm, fe_params, platform_params, + dir, SOF_WIDGET_UNPREPARE); return ret; } diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 4a62ccc71fcb..9f3a038fe21a 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1388,14 +1388,15 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret < 0) { dev_err(scomp->dev, "failed to parse component pin tokens for %s\n", w->name); - return ret; + goto widget_free; } if (swidget->num_sink_pins > SOF_WIDGET_MAX_NUM_PINS || swidget->num_source_pins > SOF_WIDGET_MAX_NUM_PINS) { dev_err(scomp->dev, "invalid pins for %s: [sink: %d, src: %d]\n", swidget->widget->name, swidget->num_sink_pins, swidget->num_source_pins); - return -EINVAL; + ret = -EINVAL; + goto widget_free; } if (swidget->num_sink_pins > 1) { @@ -1404,7 +1405,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret < 0) { dev_err(scomp->dev, "failed to parse sink pin binding for %s\n", w->name); - return ret; + goto widget_free; } } @@ -1414,7 +1415,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret < 0) { dev_err(scomp->dev, "failed to parse source pin binding for %s\n", w->name); - return ret; + goto widget_free; } } @@ -1436,9 +1437,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, case snd_soc_dapm_dai_out: dai = kzalloc(sizeof(*dai), GFP_KERNEL); if (!dai) { - kfree(swidget); - return -ENOMEM; - + ret = -ENOMEM; + goto widget_free; } ret = sof_widget_parse_tokens(scomp, swidget, tw, token_list, token_list_size); @@ -1496,8 +1496,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, tw->shift, swidget->id, tw->name, strnlen(tw->sname, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) > 0 ? tw->sname : "none"); - kfree(swidget); - return ret; + goto widget_free; } if (sof_debug_check_flag(SOF_DBG_DISABLE_MULTICORE)) { @@ -1518,10 +1517,7 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, if (ret) { dev_err(scomp->dev, "widget event binding failed for %s\n", swidget->widget->name); - kfree(swidget->private); - kfree(swidget->tuples); - kfree(swidget); - return ret; + goto free; } } } @@ -1532,10 +1528,8 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, spipe = kzalloc(sizeof(*spipe), GFP_KERNEL); if (!spipe) { - kfree(swidget->private); - kfree(swidget->tuples); - kfree(swidget); - return -ENOMEM; + ret = -ENOMEM; + goto free; } spipe->pipe_widget = swidget; @@ -1546,6 +1540,12 @@ static int sof_widget_ready(struct snd_soc_component *scomp, int index, w->dobj.private = swidget; list_add(&swidget->list, &sdev->widget_list); return ret; +free: + kfree(swidget->private); + kfree(swidget->tuples); +widget_free: + kfree(swidget); + return ret; } static int sof_route_unload(struct snd_soc_component *scomp, diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 1e0af1179ca8..25563a5093c5 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -455,8 +455,8 @@ static void push_back_to_ready_list(struct snd_usb_endpoint *ep, * This function is used both for implicit feedback endpoints and in low- * latency playback mode. */ -void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, - bool in_stream_lock) +int snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, + bool in_stream_lock) { bool implicit_fb = snd_usb_endpoint_implicit_feedback_sink(ep); @@ -480,7 +480,7 @@ void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, spin_unlock_irqrestore(&ep->lock, flags); if (ctx == NULL) - return; + break; /* copy over the length information */ if (implicit_fb) { @@ -495,11 +495,14 @@ void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, break; if (err < 0) { /* push back to ready list again for -EAGAIN */ - if (err == -EAGAIN) + if (err == -EAGAIN) { push_back_to_ready_list(ep, ctx); - else + break; + } + + if (!in_stream_lock) notify_xrun(ep); - return; + return -EPIPE; } err = usb_submit_urb(ctx->urb, GFP_ATOMIC); @@ -507,13 +510,16 @@ void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, usb_audio_err(ep->chip, "Unable to submit urb #%d: %d at %s\n", ctx->index, err, __func__); - notify_xrun(ep); - return; + if (!in_stream_lock) + notify_xrun(ep); + return -EPIPE; } set_bit(ctx->index, &ep->active_mask); atomic_inc(&ep->submitted_urbs); } + + return 0; } /* diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 924f4351588c..c09f68ce08b1 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -52,7 +52,7 @@ int snd_usb_endpoint_implicit_feedback_sink(struct snd_usb_endpoint *ep); int snd_usb_endpoint_next_packet_size(struct snd_usb_endpoint *ep, struct snd_urb_ctx *ctx, int idx, unsigned int avail); -void snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, - bool in_stream_lock); +int snd_usb_queue_pending_output_urbs(struct snd_usb_endpoint *ep, + bool in_stream_lock); #endif /* __USBAUDIO_ENDPOINT_H */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 405dc0bf6678..4b1c5ba121f3 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -39,8 +39,12 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, case UAC_VERSION_1: default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; - if (format >= 64) - return 0; /* invalid format */ + if (format >= 64) { + usb_audio_info(chip, + "%u:%d: invalid format type 0x%llx is detected, processed as PCM\n", + fp->iface, fp->altsetting, format); + format = UAC_FORMAT_TYPE_I_PCM; + } sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; format = 1ULL << format; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d959da7a1afb..eec5232f9fb2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1639,7 +1639,7 @@ static int snd_usb_pcm_playback_ack(struct snd_pcm_substream *substream) * outputs here */ if (!ep->active_mask) - snd_usb_queue_pending_output_urbs(ep, true); + return snd_usb_queue_pending_output_urbs(ep, true); return 0; } |