November 2021 - Alsa-devel - mailman.alsa-project.org

ASoC: mt8192: remove unnecessary CONFIG_PM
by Jiaxin Yu 23 Nov '21

23 Nov '21

[PATCH] ASoC: mt8192: remove unnecessary CONFIG_PM

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Re: [PATCH] ALSA: hda/via - fix occasionally no sound with VT1802
by Takashi Iwai 23 Nov '21

23 Nov '21

On Tue, 23 Nov 2021 09:07:14 +0100, tianyucode wrote: > > From: tianyu <tianyu(a)kylinos.cn> > > Hi guys, I'm a new guy for audio driver. > > I have encountered a problem. When using the VT1802 codec, > there will be no sound occasionally. After checking the initialization process, > I found that the cause of the problem is that the power state does not have > hda_set_power_state(codec, AC_PWRST_D0). > > 1. The power state is set to AC_PWRST_D0 during the first initialization： > > azx_probe_continue > azx_probe_codecs > snd_hda_codec_new > snd_hda_codec_device_new > hda_set_power_state(codec, AC_PWRST_D0); > > 2. Then, the power state is set to AC_PWRST_D3 in VT1802, but not set in realtek (ALC269). > > azx_probe_continue > azx_codec_configure > … > patch_vt2002 > via_parse_auto_config > snd_hda_gen_parse_auto_config > add_all_pin_power_ctls > add_pin_power_ctls > set_path_power > > static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) > { > /**/ > if (!codec->power_save_node) > return; > > add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); > /**/ > } > > The power_save_node is set to 0 in patch_alc269 and it returns here, > but it is set to 1 in VT1802. > > patch_vt2002P > via_new_spec > codec->power_save_node = 1; > via_parse_auto_config > snd_hda_gen_parse_auto_config > codec->power_save_node = 0;//it set to 0 here,but add_all_pin_power_ctls has been done > > patch_alc269 > codec->power_save_node = 0; > > 3. Next there is a suspend and resume process, the resume process will also set the > power state to AC_PWRST_D0, but I think there is a problem with this process. > > (1)suspend: > > azx_probe_continue > snd_card_register > .. > snd_hda_codec_dev_register > snd_hda_codec_register > snd_hda_power_down > pm_runtime_put_autosuspend > __pm_runtime_suspend(dev, RPM_GET_PUT | RPM_ASYNC | RPM_AUTO); > rpm_suspend > > static int rpm_suspend(struct device *dev, int rpmflags) > { > /**/ > if (rpmflags & RPM_ASYNC) { > dev->power.request = (rpmflags & RPM_AUTO) ? > RPM_REQ_AUTOSUSPEND : RPM_REQ_SUSPEND; > if (!dev->power.request_pending) { > dev->power.request_pending = true; > queue_work(pm_wq, &dev->power.work); > } > goto out; > } > > __update_runtime_status(dev, RPM_SUSPENDING); > > /**/ > retval = rpm_callback(callback, dev); > __update_runtime_status(dev, RPM_SUSPENDED); > /**/ > } > > The first call to rpm_suspend has the RPM_ASYNC flag, so queue_work is called, > and the work queue is used to enter rpm_suspend > > pm_runtime_init > INIT_WORK(&dev->power.work, pm_runtime_work); > > (2)resume: > > azx_probe_continue > set_default_power_save > snd_hda_set_power_save(&chip->bus, val * 1000); > codec_set_power_save(c, delay); > > int val = power_save; > static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;//It is 0 in my .config > > static void codec_set_power_save(struct hda_codec *codec, int delay) > { > struct device *dev = hda_codec_dev(codec); > if (delay == 0 && codec->auto_runtime_pm) > delay = 3000; > > if (delay > 0) { > pm_runtime_set_autosuspend_delay(dev, delay); > pm_runtime_use_autosuspend(dev); > pm_runtime_allow(dev); > if (!pm_runtime_suspended(dev)) > pm_runtime_mark_last_busy(dev); > } else { > pm_runtime_dont_use_autosuspend(dev); > pm_runtime_forbid(dev); > } > } > > pm_runtime_forbid > rpm_resume(dev, 0); > > static int rpm_resume(struct device *dev, int rpmflags) > { > /**/ > if (dev->power.runtime_status == RPM_ACTIVE) { > retval = 1; > goto out; > } > > /**/ > _update_runtime_status(dev, RPM_RESUMING); > retval = rpm_callback(callback, dev); > /**/ > __update_runtime_status(dev, RPM_ACTIVE); > /**/ > } > > The callback functions of suspend and resume are as follows, which set the power state: > > hda_call_codec_suspend > state = hda_set_power_state(codec, AC_PWRST_D3); > > hda_call_codec_resume > hda_set_power_state(codec, AC_PWRST_D0); > > You can see that the resume function relies on dev->power.runtime_status, > and the status is set in rpm_suspend. The operation of rpm_suspend depends > on the scheduling of the work queue. I added print debugging on my machine, > and occasionally there will be pm_runtime_work in Run after rpm_resume. > At this time, the suspend and resume processes are not performed correctly. > In VT1802, the power_state is still D3, and the machine has no sound. > > 4. I searched the Internet and did not find the relevant modification, > but found the commit 317d9313925c (ALSA: hda/realtek-Set default power save node to 0). > > Does VIA need to be modified like this? > > Since I am not familiar with ALSA and runtime pm, so come here to consult. For VIA codecs, there is an additional fixup table to turn off the power_save_node. Add an entry with the PCI SSID of your device to vt2002p_fixups[] table with VIA_FIXUP_POWER_SAVE. thanks, Takashi

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[PATCH] ALSA: hda/via - fix occasionally no sound with VT1802
by tianyucode 23 Nov '21

23 Nov '21

From: tianyu <tianyu(a)kylinos.cn> Hi guys, I'm a new guy for audio driver. I have encountered a problem. When using the VT1802 codec, there will be no sound occasionally. After checking the initialization process, I found that the cause of the problem is that the power state does not have hda_set_power_state(codec, AC_PWRST_D0). 1. The power state is set to AC_PWRST_D0 during the first initialization： azx_probe_continue azx_probe_codecs snd_hda_codec_new snd_hda_codec_device_new hda_set_power_state(codec, AC_PWRST_D0); 2. Then, the power state is set to AC_PWRST_D3 in VT1802, but not set in realtek (ALC269). azx_probe_continue azx_codec_configure … patch_vt2002 via_parse_auto_config snd_hda_gen_parse_auto_config add_all_pin_power_ctls add_pin_power_ctls set_path_power static void add_all_pin_power_ctls(struct hda_codec *codec, bool on) { /**/ if (!codec->power_save_node) return; add_pin_power_ctls(codec, cfg->line_outs, cfg->line_out_pins, on); /**/ } The power_save_node is set to 0 in patch_alc269 and it returns here, but it is set to 1 in VT1802. patch_vt2002P via_new_spec codec->power_save_node = 1; via_parse_auto_config snd_hda_gen_parse_auto_config codec->power_save_node = 0;//it set to 0 here,but add_all_pin_power_ctls has been done patch_alc269 codec->power_save_node = 0; 3. Next there is a suspend and resume process, the resume process will also set the power state to AC_PWRST_D0, but I think there is a problem with this process. (1)suspend: azx_probe_continue snd_card_register .. snd_hda_codec_dev_register snd_hda_codec_register snd_hda_power_down pm_runtime_put_autosuspend __pm_runtime_suspend(dev, RPM_GET_PUT | RPM_ASYNC | RPM_AUTO); rpm_suspend static int rpm_suspend(struct device *dev, int rpmflags) { /**/ if (rpmflags & RPM_ASYNC) { dev->power.request = (rpmflags & RPM_AUTO) ? RPM_REQ_AUTOSUSPEND : RPM_REQ_SUSPEND; if (!dev->power.request_pending) { dev->power.request_pending = true; queue_work(pm_wq, &dev->power.work); } goto out; } __update_runtime_status(dev, RPM_SUSPENDING); /**/ retval = rpm_callback(callback, dev); __update_runtime_status(dev, RPM_SUSPENDED); /**/ } The first call to rpm_suspend has the RPM_ASYNC flag, so queue_work is called, and the work queue is used to enter rpm_suspend pm_runtime_init INIT_WORK(&dev->power.work, pm_runtime_work); (2)resume: azx_probe_continue set_default_power_save snd_hda_set_power_save(&chip->bus, val * 1000); codec_set_power_save(c, delay); int val = power_save; static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;//It is 0 in my .config static void codec_set_power_save(struct hda_codec *codec, int delay) { struct device *dev = hda_codec_dev(codec); if (delay == 0 && codec->auto_runtime_pm) delay = 3000; if (delay > 0) { pm_runtime_set_autosuspend_delay(dev, delay); pm_runtime_use_autosuspend(dev); pm_runtime_allow(dev); if (!pm_runtime_suspended(dev)) pm_runtime_mark_last_busy(dev); } else { pm_runtime_dont_use_autosuspend(dev); pm_runtime_forbid(dev); } } pm_runtime_forbid rpm_resume(dev, 0); static int rpm_resume(struct device *dev, int rpmflags) { /**/ if (dev->power.runtime_status == RPM_ACTIVE) { retval = 1; goto out; } /**/ _update_runtime_status(dev, RPM_RESUMING); retval = rpm_callback(callback, dev); /**/ __update_runtime_status(dev, RPM_ACTIVE); /**/ } The callback functions of suspend and resume are as follows, which set the power state: hda_call_codec_suspend state = hda_set_power_state(codec, AC_PWRST_D3); hda_call_codec_resume hda_set_power_state(codec, AC_PWRST_D0); You can see that the resume function relies on dev->power.runtime_status, and the status is set in rpm_suspend. The operation of rpm_suspend depends on the scheduling of the work queue. I added print debugging on my machine, and occasionally there will be pm_runtime_work in Run after rpm_resume. At this time, the suspend and resume processes are not performed correctly. In VT1802, the power_state is still D3, and the machine has no sound. 4. I searched the Internet and did not find the relevant modification, but found the commit 317d9313925c (ALSA: hda/realtek-Set default power save node to 0). Does VIA need to be modified like this? Since I am not familiar with ALSA and runtime pm, so come here to consult. --- sound/pci/hda/patch_via.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index a29d7e989fe6..d01ea582cf25 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1073,8 +1073,10 @@ static int patch_vt2002P(struct hda_codec *codec) snd_hda_pick_fixup(codec, NULL, vt2002p_fixups, via_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - if (spec->codec_type == VT1802) + if (spec->codec_type == VT1802) { + codec->power_save_node = 0; err = snd_hda_add_verbs(codec, vt1802_init_verbs); + } else err = snd_hda_add_verbs(codec, vt2002P_init_verbs); if (err < 0) -- 2.25.1

1 0

[RESEND,v3] ASoC: Intel: boards: add max98390 2/4 speakers support
by Mac Chiang 23 Nov '21

23 Nov '21

support 2 hw boards. 1. SSP2 connects max98390, 2 speakers. 2. SSP1 connects max98390, 2/4 speakers. 2 or 4 speakers playback add echo reference capture add bt offload support add DMI_OEM_STRING for board variants add ALC5682I-VS support Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Kieth Tzeng <keith.tzeng(a)quantatw.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- changes in resend v3: - use .comp_ids to scan acpi support ALC5682I-VD or ALC5682I-VS --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 180 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 16 ++ sound/soc/intel/boards/sof_rt5682.c | 84 ++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 21 ++ sound/soc/sof/sof-pci-dev.c | 9 + 6 files changed, 311 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 849445fcc05d..34ccefcc30c7 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -467,6 +467,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98390 select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e66dfe666915..9171d9cd179e 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -5,6 +5,7 @@ #include <linux/string.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -133,6 +134,185 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ +const struct snd_soc_dapm_route max_98390_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; + +static const struct snd_kcontrol_new max_98390_tt_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("TL Spk"), + SOC_DAPM_PIN_SWITCH("TR Spk"), +}; + +static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { + SND_SOC_DAPM_SPK("TL Spk", NULL), + SND_SOC_DAPM_SPK("TR Spk", NULL), +}; + +const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { + /* Tweeter speaker */ + { "TL Spk", NULL, "Tweeter Left BE_OUT" }, + { "TR Spk", NULL, "Tweeter Right BE_OUT" }, +}; + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +static struct snd_soc_codec_conf max_98390_4spk_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV2_NAME), + .name_prefix = "Tweeter Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV3_NAME), + .name_prefix = "Tweeter Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_dai_link_component max_98390_4spk_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV2_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV3_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_4spk_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + if (i >= ARRAY_SIZE(max_98390_4spk_components)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", i); + return -ENODEV; + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x01, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x02, 3, 4, 32); + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV2_NAME)) { + /* DEVi2 tdm slot configuration Tweeter Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x04, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV3_NAME)) { + /* DEV3 tdm slot configuration Tweeter Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x08, 3, 4, 32); + } + } + return 0; +} + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + /* add regular speakers dapm route */ + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_dapm_routes, + ARRAY_SIZE(max_98390_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "unable to add Left/Right Speaker dapm, ret %d\n", ret); + return ret; + } + + /* add widgets/controls/dapm for tweeter speakers */ + if (acpi_dev_present("MX98390", "3", -1)) { + ret = snd_soc_dapm_new_controls(&card->dapm, max_98390_tt_dapm_widgets, + ARRAY_SIZE(max_98390_tt_dapm_widgets)); + + if (ret) { + dev_err(rtd->dev, "unable to add tweeter dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, max_98390_tt_kcontrols, + ARRAY_SIZE(max_98390_tt_kcontrols)); + if (ret) { + dev_err(rtd->dev, "unable to add tweeter card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_tt_dapm_routes, + ARRAY_SIZE(max_98390_tt_dapm_routes)); + if (ret) + dev_err(rtd->dev, + "unable to add Tweeter Left/Right Speaker dapm, ret %d\n", ret); + } + return ret; +} +EXPORT_SYMBOL_NS(max_98390_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +const struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch) +{ + if (ch == ARRAY_SIZE(max_98390_4spk_codec_conf)) { + card->codec_conf = max_98390_4spk_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_4spk_codec_conf); + } else { + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); + } +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 3ff5e8fec4de..7a8c53049e4d 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,22 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" +#define MAX_98390_DEV2_NAME "i2c-MX98390:02" +#define MAX_98390_DEV3_NAME "i2c-MX98390:03" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_dai_link_component max_98390_4spk_components[4]; +extern const struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch); +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index c41f386b4138..3869b38274b9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -59,6 +59,9 @@ (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) #define SOF_SSP_BT_OFFLOAD_PRESENT BIT(22) #define SOF_RT5682S_HEADPHONE_CODEC_PRESENT BIT(23) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) +#define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) + /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -179,6 +182,36 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S_4SPK"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_MAX98390_TWEETER_SPEAKER_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, {} }; @@ -486,6 +519,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (err < 0) return err; } + return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -784,6 +818,20 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & + SOF_MAX98390_TWEETER_SPEAKER_PRESENT) { + links[id].codecs = max_98390_4spk_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_4spk_components); + } else { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + } + links[id].init = max_98390_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; + } else { max_98357a_dai_link(&links[id]); } @@ -868,6 +916,10 @@ static int sof_audio_probe(struct platform_device *pdev) if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + /* Detect the headset codec variant to support machines in DMI quirk */ + if (acpi_dev_present("RTL5682", NULL, -1)) + sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -924,6 +976,14 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) { + if (sof_rt5682_quirk & SOF_MAX98390_TWEETER_SPEAKER_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_4spk_components)); + else + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_components)); + } if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1050,6 +1110,29 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { + .name = "adl_mx98390_rt5682s", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_RT5682S_HEADPHONE_CODEC_PRESENT | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { .name = "adl_mx98360_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | @@ -1080,6 +1163,7 @@ MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); 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 fde310e5724b..0d04166b9d64 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -379,6 +379,11 @@ static const struct snd_soc_acpi_codecs adl_rt1019p_amp = { .codecs = {"RTL1019"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .comp_ids = &adl_rt5682_rt5682s_hp, @@ -434,6 +439,22 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-nau8825.tplg", }, + { + .comp_ids = &adl_rt5682_rt5682s_hp, + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, + { + .comp_ids = &adl_rt5682_rt5682s_hp, + .drv_name = "adl_mx98390_rt5682s", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..be90a5f79766 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -59,6 +59,15 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-rt5682-ssp0-max98373-ssp2.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = "sof-adl-max98390-ssp2-rt5682-ssp0.tplg", + }, + {} }; -- 2.20.1

3 2

[PATCH] ASoC: qcom: Add support for ALC5682I-VS codec
by lvzhaoxiong 23 Nov '21

23 Nov '21

Qcom machine driver adds rt5682s support in this patch. Card name can be specified from dts by model property, and driver makes use of the name to distinguish which headset codec is on the board. Signed-off-by: lvzhaoxiong <lvzhaoxiong(a)huaqin.corp-partner.google.com> --- sound/soc/qcom/Kconfig | 1 + sound/soc/qcom/sc7180.c | 24 +++++++++++++++++++----- 2 files changed, 20 insertions(+), 5 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index b2173847dc47..cf3e151bb635 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -169,6 +169,7 @@ config SND_SOC_SC7180 select SND_SOC_LPASS_SC7180 select SND_SOC_MAX98357A select SND_SOC_RT5682_I2C + select SND_SOC_RT5682S select SND_SOC_ADAU7002 help To add support for audio on Qualcomm Technologies Inc. diff --git a/sound/soc/qcom/sc7180.c b/sound/soc/qcom/sc7180.c index 768566bb57a5..2fff764a00a7 100644 --- a/sound/soc/qcom/sc7180.c +++ b/sound/soc/qcom/sc7180.c @@ -17,6 +17,7 @@ #include <uapi/linux/input-event-codes.h> #include "../codecs/rt5682.h" +#include "../codecs/rt5682s.h" #include "common.h" #include "lpass.h" @@ -128,7 +129,21 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) struct sc7180_snd_data *data = snd_soc_card_get_drvdata(card); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; + int pll_id, pll_source, pll_in, pll_out, clk_id, ret; + + if (!(strcmp(card->name, "sc7180-rt5682-max98357a-1mic"))) { + pll_source = RT5682_PLL1_S_MCLK; + pll_id = 0; + clk_id = RT5682_SCLK_S_PLL1; + pll_out = RT5682_PLL1_FREQ; + pll_in = DEFAULT_MCLK_RATE; + } else if (!(strcmp(card->name, "sc7180-rt5682s-max98357a-1mic"))) { + pll_source = RT5682S_PLL_S_MCLK; + pll_id = RT5682S_PLL2; + clk_id = RT5682S_SCLK_S_PLL2; + pll_out = RT5682_PLL1_FREQ; + pll_in = DEFAULT_MCLK_RATE; + } switch (cpu_dai->id) { case MI2S_PRIMARY: @@ -145,16 +160,15 @@ static int sc7180_snd_startup(struct snd_pcm_substream *substream) SND_SOC_DAIFMT_I2S); /* Configure PLL1 for codec */ - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, - DEFAULT_MCLK_RATE, RT5682_PLL1_FREQ); + ret = snd_soc_dai_set_pll(codec_dai, pll_id, pll_source, + pll_in, pll_out); if (ret) { dev_err(rtd->dev, "can't set codec pll: %d\n", ret); return ret; } /* Configure sysclk for codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, - RT5682_PLL1_FREQ, + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, pll_out, SND_SOC_CLOCK_IN); if (ret) dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", -- 2.17.1

1 0

[PATCH] ASoC: rk817: Add module alias for rk817-codec
by Nicolas Frattaroli 23 Nov '21

23 Nov '21

Without a module alias, autoloading the driver does not occurr when it is built as a module. By adding a module alias, the driver now probes fine automatically and therefore analog audio output works as it should. Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Nicolas Frattaroli <frattaroli.nicolas(a)gmail.com> --- sound/soc/codecs/rk817_codec.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 943d7d933e81..03f24edfe4f6 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -539,3 +539,4 @@ module_platform_driver(rk817_codec_driver); MODULE_DESCRIPTION("ASoC RK817 codec driver"); MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:rk817-codec"); -- 2.34.0

3 2

[PATCH] ASoC: SOF: sof-pci-dev: use community key on all Up boards
by Pierre-Louis Bossart 23 Nov '21

23 Nov '21

There are already 3 versions of the Up boards with support for the SOF community key (ApolloLake, WhiskyLake, TigerLake). Rather than continue to add quirks for each version, let's add a wildcard. For WHL and TGL, the authentication supports both the SOF community key and the firmware signed with the Intel production key. Given two choices, the community key is the preferred option to allow developers to sign their own firmware. The firmware signed with production key can still be selected if needed with a kernel module option (snd-sof-pci.fw_path="intel/sof") Tested-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/sof-pci-dev.c | 10 +--------- 1 file changed, 1 insertion(+), 9 deletions(-) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..b4bc4f887b43 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -64,17 +64,9 @@ static const struct dmi_system_id sof_tplg_table[] = { static const struct dmi_system_id community_key_platforms[] = { { - .ident = "Up Squared", + .ident = "Up boards", .matches = { DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), - DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), - } - }, - { - .ident = "Up Extreme", - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), - DMI_MATCH(DMI_BOARD_NAME, "UP-WHL01"), } }, { -- 2.25.1

2 1

[PATCH 00/10] ASoC: SOF: enable multicore with dynamic pipelines
by Kai Vehmanen 23 Nov '21

23 Nov '21

Hi, initial support for SOF dynamic pipelines was added in commit 5fcdbb2d45df ("ASoC: SOF: Add support for dynamic pipelines"). When a pipeline is marked dynamic in the SOF DSP firmware topology definition (the tplg file kernel loads from filesystem), it means the pipeline resources are not allocated when DSP is booted (at driver probe, or at runtime resume), but rather delayed until the pipeline is actually used. Until now, it has not been possible to use multiple DSP cores in a topology that uses dynamic pipeline capability. This patchset addresses this gap and allows to freely mix dynamic pipelines and multicore usages in topology definitions. The series was originally reviewed in https://github.com/thesofproject/linux/pull/3019 Ranjani Sridharan (10): ASoC: SOF: Intel: hda: expose get_chip_info() ASoC: SOF: Introduce num_cores and ref count per core ASoC: SOF: Add ops for core_get and core_put ASoC: SOF: Intel: TGL: set core_get/put ops ASoC: SOF: Intel: CNL/ICL/APL: set core_get/core_put ops ASoC: SOF: topology: remove sof_load_pipeline_ipc() ASoC: SOF: free widgets in sof_tear_down_pipelines() for static pipelines ASoC: SOF: hda: don't use the core op for power up/power down ASoC: SOF: add support for dynamic pipelines with multi-core ASoC: SOF: Intel: hda: free DAI widget during stop and suspend sound/soc/sof/imx/imx8.c | 1 + sound/soc/sof/imx/imx8m.c | 1 + sound/soc/sof/intel/apl.c | 5 +- sound/soc/sof/intel/bdw.c | 9 +++ sound/soc/sof/intel/byt.c | 9 +++ sound/soc/sof/intel/cnl.c | 5 +- sound/soc/sof/intel/hda-dai.c | 23 +------ sound/soc/sof/intel/hda-dsp.c | 52 +++++++++++++++- sound/soc/sof/intel/hda-loader.c | 24 +++++--- sound/soc/sof/intel/hda.c | 17 ++---- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/icl.c | 5 +- sound/soc/sof/intel/pci-tng.c | 9 +++ sound/soc/sof/intel/shim.h | 7 +++ sound/soc/sof/intel/tgl.c | 46 +++++++++++++- sound/soc/sof/ops.h | 71 ++++++++++++++++------ sound/soc/sof/sof-audio.c | 83 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 6 -- sound/soc/sof/sof-priv.h | 21 +++++-- sound/soc/sof/topology.c | 100 ------------------------------- 20 files changed, 294 insertions(+), 201 deletions(-) base-commit: 626a3dfbdb5d11f92e709680135abf272057ef59 -- 2.33.0

2 11

[PATCH] ASoC: cs42l42: Report initial jack state
by Richard Fitzgerald 23 Nov '21

23 Nov '21

When a jack handler is registered in cs42l42_set_jack() the initial state should be reported if an attached headphone/headset has already been detected. The jack detect sequence takes around 1 second: typically long enough for the machine driver to probe and register the jack handler in time to receive the first report from the interrupt handler. So it is possible on some systems that the correct initial state was seen simply because of lucky timing. Modular builds were more likely to miss the reporting of the initial state. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 4ca239f33737 ("ASoC: cs42l42: Always enable TS_PLUG and TS_UNPLUG interrupts") --- sound/soc/codecs/cs42l42.c | 22 ++++++++++++++++++++++ sound/soc/codecs/cs42l42.h | 2 ++ 2 files changed, 24 insertions(+) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 0c4303547fd8..43d98bdb5b5b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -549,8 +549,25 @@ static int cs42l42_set_jack(struct snd_soc_component *component, struct snd_soc_ { struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + /* Prevent race with interrupt handler */ + mutex_lock(&cs42l42->jack_detect_mutex); cs42l42->jack = jk; + if (jk) { + switch (cs42l42->hs_type) { + case CS42L42_PLUG_CTIA: + case CS42L42_PLUG_OMTP: + snd_soc_jack_report(jk, SND_JACK_HEADSET, SND_JACK_HEADSET); + break; + case CS42L42_PLUG_HEADPHONE: + snd_soc_jack_report(jk, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); + break; + default: + break; + } + } + mutex_unlock(&cs42l42->jack_detect_mutex); + return 0; } @@ -1618,6 +1635,8 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) CS42L42_M_DETECT_FT_MASK | CS42L42_M_HSBIAS_HIZ_MASK); + mutex_lock(&cs42l42->jack_detect_mutex); + /* Check auto-detect status */ if ((~masks[5]) & irq_params_table[5].mask) { if (stickies[5] & CS42L42_HSDET_AUTO_DONE_MASK) { @@ -1686,6 +1705,8 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) } } + mutex_unlock(&cs42l42->jack_detect_mutex); + return IRQ_HANDLED; } @@ -2033,6 +2054,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, cs42l42->dev = &i2c_client->dev; i2c_set_clientdata(i2c_client, cs42l42); + mutex_init(&cs42l42->jack_detect_mutex); cs42l42->regmap = devm_regmap_init_i2c(i2c_client, &cs42l42_regmap); if (IS_ERR(cs42l42->regmap)) { diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 75ade987d0db..9fff183dce8e 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -12,6 +12,7 @@ #ifndef __CS42L42_H__ #define __CS42L42_H__ +#include <linux/mutex.h> #include <sound/jack.h> #define CS42L42_PAGE_REGISTER 0x00 /* Page Select Register */ @@ -841,6 +842,7 @@ struct cs42l42_private { struct gpio_desc *reset_gpio; struct completion pdn_done; struct snd_soc_jack *jack; + struct mutex jack_detect_mutex; int pll_config; int bclk; u32 sclk; -- 2.11.0

2 1

[PATCH v2 0/3] ASoC: stm32: add pm runtime support
by Olivier Moysan 23 Nov '21

23 Nov '21

Enable support of pm runtime on STM32 SPDIFRX, I2S and DFSDM drivers to allow power state monitoring. Changes in v2: - Move pm runtime enabling before component registration Olivier Moysan (3): ASoC: stm32: i2s: add pm_runtime support ASoC: stm32: dfsdm: add pm_runtime support for audio ASoC: stm32: spdifrx: add pm_runtime support sound/soc/stm/stm32_adfsdm.c | 5 ++++- sound/soc/stm/stm32_i2s.c | 4 ++++ sound/soc/stm/stm32_spdifrx.c | 4 ++++ 3 files changed, 12 insertions(+), 1 deletion(-) -- 2.17.1

2 4