diff options
Diffstat (limited to 'sound')
43 files changed, 585 insertions, 482 deletions
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ec4536c8d8d4..dafcf82139e2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -932,7 +932,7 @@ int snd_hda_bus_new(struct snd_card *card, } EXPORT_SYMBOL_GPL(snd_hda_bus_new); -#ifdef CONFIG_SND_HDA_GENERIC +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) #define is_generic_config(codec) \ (codec->modelname && !strcmp(codec->modelname, "generic")) #else @@ -1339,23 +1339,15 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) /* * Dynamic symbol binding for the codec parsers */ -#ifdef MODULE -#define load_parser_sym(sym) ((int (*)(struct hda_codec *))symbol_request(sym)) -#define unload_parser_addr(addr) symbol_put_addr(addr) -#else -#define load_parser_sym(sym) (sym) -#define unload_parser_addr(addr) do {} while (0) -#endif #define load_parser(codec, sym) \ - ((codec)->parser = load_parser_sym(sym)) + ((codec)->parser = (int (*)(struct hda_codec *))symbol_request(sym)) static void unload_parser(struct hda_codec *codec) { - if (codec->parser) { - unload_parser_addr(codec->parser); - codec->parser = NULL; - } + if (codec->parser) + symbol_put_addr(codec->parser); + codec->parser = NULL; } /* @@ -1570,7 +1562,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) EXPORT_SYMBOL_GPL(snd_hda_codec_update_widgets); -#ifdef CONFIG_SND_HDA_CODEC_HDMI +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) /* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ static bool is_likely_hdmi_codec(struct hda_codec *codec) { @@ -1620,12 +1612,20 @@ int snd_hda_codec_configure(struct hda_codec *codec) patch = codec->preset->patch; if (!patch) { unload_parser(codec); /* to be sure */ - if (is_likely_hdmi_codec(codec)) + if (is_likely_hdmi_codec(codec)) { +#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) patch = load_parser(codec, snd_hda_parse_hdmi_codec); -#ifdef CONFIG_SND_HDA_GENERIC - if (!patch) +#elif IS_BUILTIN(CONFIG_SND_HDA_CODEC_HDMI) + patch = snd_hda_parse_hdmi_codec; +#endif + } + if (!patch) { +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) patch = load_parser(codec, snd_hda_parse_generic_codec); +#elif IS_BUILTIN(CONFIG_SND_HDA_GENERIC) + patch = snd_hda_parse_generic_codec; #endif + } if (!patch) { printk(KERN_ERR "hda-codec: No codec parser is available\n"); return -ENODEV; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8321a97d5c05..d9a09bdd09db 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3269,7 +3269,7 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->control_mutex); snd_hda_codec_flush_cache(codec); /* flush the updates */ if (err >= 0 && spec->cap_sync_hook) - spec->cap_sync_hook(codec, ucontrol); + spec->cap_sync_hook(codec, kcontrol, ucontrol); return err; } @@ -3390,7 +3390,7 @@ static int cap_single_sw_put(struct snd_kcontrol *kcontrol, return ret; if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, ucontrol); + spec->cap_sync_hook(codec, kcontrol, ucontrol); return ret; } @@ -3795,7 +3795,7 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; snd_hda_activate_path(codec, path, true, false); if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, NULL); + spec->cap_sync_hook(codec, NULL, NULL); path_power_down_sync(codec, old_path); return 1; } @@ -5270,7 +5270,7 @@ static void init_input_src(struct hda_codec *codec) } if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, NULL); + spec->cap_sync_hook(codec, NULL, NULL); } /* set right pin controls for digital I/O */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 07f767231c9f..c908afbe4d94 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -274,6 +274,7 @@ struct hda_gen_spec { void (*init_hook)(struct hda_codec *codec); void (*automute_hook)(struct hda_codec *codec); void (*cap_sync_hook)(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /* PCM hooks */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a21a50..e354ab1ec20f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -198,7 +198,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) -#ifdef CONFIG_SND_HDA_CODEC_HDMI +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) #define SUPPORT_VGA_SWITCHEROO #endif #endif diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 7a426ed491f2..8ed0bcc01386 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -244,6 +244,19 @@ static void ad_fixup_inv_jack_detect(struct hda_codec *codec, } } +/* Toshiba Satellite L40 implements EAPD in a standard way unlike others */ +static void ad1986a_fixup_eapd(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + codec->inv_eapd = 0; + spec->gen.keep_eapd_on = 1; + spec->eapd_nid = 0x1b; + } +} + enum { AD1986A_FIXUP_INV_JACK_DETECT, AD1986A_FIXUP_ULTRA, @@ -251,6 +264,7 @@ enum { AD1986A_FIXUP_3STACK, AD1986A_FIXUP_LAPTOP, AD1986A_FIXUP_LAPTOP_IMIC, + AD1986A_FIXUP_EAPD, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -311,6 +325,10 @@ static const struct hda_fixup ad1986a_fixups[] = { .chained_before = 1, .chain_id = AD1986A_FIXUP_LAPTOP, }, + [AD1986A_FIXUP_EAPD] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1986a_fixup_eapd, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { @@ -318,6 +336,7 @@ static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK), + SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA), @@ -472,6 +491,8 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec) static int patch_ad1983(struct hda_codec *codec) { struct ad198x_spec *spec; + static hda_nid_t conn_0c[] = { 0x08 }; + static hda_nid_t conn_0d[] = { 0x09 }; int err; err = alloc_ad_spec(codec); @@ -479,8 +500,14 @@ static int patch_ad1983(struct hda_codec *codec) return err; spec = codec->spec; + spec->gen.mixer_nid = 0x0e; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + + /* limit the loopback routes not to confuse the parser */ + snd_hda_override_conn_list(codec, 0x0c, ARRAY_SIZE(conn_0c), conn_0c); + snd_hda_override_conn_list(codec, 0x0d, ARRAY_SIZE(conn_0d), conn_0d); + err = ad198x_parse_auto_config(codec, false); if (err < 0) goto error; @@ -999,6 +1026,9 @@ static void ad1884_fixup_thinkpad(struct hda_codec *codec, spec->gen.keep_eapd_on = 1; spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; spec->eapd_nid = 0x12; + /* Analog PC Beeper - allow firmware/ACPI beeps */ + spec->beep_amp = HDA_COMPOSE_AMP_VAL(0x20, 3, 3, HDA_INPUT); + spec->gen.beep_nid = 0; /* no digital beep */ } } @@ -1065,6 +1095,7 @@ static int patch_ad1884(struct hda_codec *codec) spec = codec->spec; spec->gen.mixer_nid = 0x20; + spec->gen.mixer_merge_nid = 0x21; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 54d14793725a..46ecdbb9053f 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2662,60 +2662,6 @@ static bool dspload_wait_loaded(struct hda_codec *codec) } /* - * PCM stuffs - */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) -{ - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd( - "ca0132_setup_stream: NID=0x%x, stream=0x%x, " - "channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int val; - - if (!nid) - return; - - snd_printdd(KERN_INFO "ca0132_cleanup_stream: NID=0x%x\n", nid); - - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (!val) - return; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); -} - -/* * PCM callbacks */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -2726,7 +2672,7 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, { struct ca0132_spec *spec = codec->spec; - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); return 0; } @@ -2745,7 +2691,7 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) msleep(50); - ca0132_cleanup_stream(codec, spec->dacs[0]); + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); return 0; } @@ -2822,10 +2768,8 @@ static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, unsigned int format, struct snd_pcm_substream *substream) { - struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, hinfo->nid, + stream_tag, 0, format); return 0; } @@ -2839,7 +2783,7 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->dsp_state == DSP_DOWNLOADING) return 0; - ca0132_cleanup_stream(codec, hinfo->nid); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } @@ -4742,6 +4686,8 @@ static int patch_ca0132(struct hda_codec *codec) return err; codec->patch_ops = ca0132_patch_ops; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; return 0; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4e0ec146553d..bcf91bea3317 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3291,7 +3291,8 @@ static void cxt_update_headset_mode(struct hda_codec *codec) } static void cxt_update_headset_mode_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { cxt_update_headset_mode(codec); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56a8f1876603..850296a1e0ff 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -708,7 +708,8 @@ static void alc_inv_dmic_sync(struct hda_codec *codec, bool force) } static void alc_inv_dmic_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { alc_inv_dmic_sync(codec, false); } @@ -1821,6 +1822,7 @@ enum { ALC889_FIXUP_IMAC91_VREF, ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, + ALC889_FIXUP_MP11_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -2190,6 +2192,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC889_FIXUP_MBP_VREF, }, + [ALC889_FIXUP_MP11_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba11_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, @@ -2253,7 +2261,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC889_FIXUP_MP11_VREF), SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), @@ -3211,7 +3219,8 @@ static void alc269_fixup_hp_gpio_mute_hook(void *private_data, int enabled) /* turn on/off mic-mute LED per capture hook */ static void alc269_fixup_hp_gpio_mic_mute_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct alc_spec *spec = codec->spec; unsigned int oldval = spec->gpio_led; @@ -3521,7 +3530,8 @@ static void alc_update_headset_mode(struct hda_codec *codec) } static void alc_update_headset_mode_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { alc_update_headset_mode(codec); } @@ -4243,6 +4253,7 @@ static const struct hda_fixup alc269_fixups[] = { }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0283, "Acer TravelMate 8371", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), @@ -4298,7 +4309,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0657, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), @@ -4307,6 +4320,54 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), + /* ALC282 */ + SND_PCI_QUIRK(0x103c, 0x220f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2213, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2266, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2267, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2269, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22a0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c1, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22cd, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22ce, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22d0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + /* ALC290 */ + SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2261, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2262, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2280, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2281, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2282, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2289, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c5, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c6, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c3, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -4322,6 +4383,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), + SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), @@ -5096,12 +5158,13 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), - SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6998cf29b9bc..3bc29c9b2529 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -83,6 +83,7 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, + STAC_92HD89XX_HP_FRONT_JACK, STAC_92HD73XX_MODELS }; @@ -97,6 +98,7 @@ enum { STAC_92HD83XXX_HP_LED, STAC_92HD83XXX_HP_INV_LED, STAC_92HD83XXX_HP_MIC_LED, + STAC_HP_LED_GPIO10, STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, @@ -194,7 +196,7 @@ struct sigmatel_spec { int default_polarity; unsigned int mic_mute_led_gpio; /* capture mute LED GPIO */ - bool mic_mute_led_on; /* current mic mute state */ + unsigned int mic_enabled; /* current mic mute state (bitmask) */ /* stream */ unsigned int stream_delay; @@ -324,19 +326,26 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, /* hook for controlling mic-mute LED GPIO */ static void stac_capture_led_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct sigmatel_spec *spec = codec->spec; - bool mute; + unsigned int mask; + bool cur_mute, prev_mute; - if (!ucontrol) + if (!kcontrol || !ucontrol) return; - mute = !(ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); - if (spec->mic_mute_led_on != mute) { - spec->mic_mute_led_on = mute; - if (mute) + mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + prev_mute = !spec->mic_enabled; + if (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]) + spec->mic_enabled |= mask; + else + spec->mic_enabled &= ~mask; + cur_mute = !spec->mic_enabled; + if (cur_mute != prev_mute) { + if (cur_mute) spec->gpio_data |= spec->mic_mute_led_gpio; else spec->gpio_data &= ~spec->mic_mute_led_gpio; @@ -1788,6 +1797,12 @@ static const struct hda_pintbl intel_dg45id_pin_configs[] = { {} }; +static const struct hda_pintbl stac92hd89xx_hp_front_jack_pin_configs[] = { + { 0x0a, 0x02214030 }, + { 0x0b, 0x02A19010 }, + {} +}; + static void stac92hd73xx_fixup_ref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -1906,6 +1921,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_NO_JD] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_no_jd, + }, + [STAC_92HD89XX_HP_FRONT_JACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = stac92hd89xx_hp_front_jack_pin_configs, } }; @@ -1966,6 +1985,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, + "unknown HP", STAC_92HD89XX_HP_FRONT_JACK), {} /* terminator */ }; @@ -2110,6 +2131,17 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, } } +static void stac92hd83xxx_fixup_hp_led_gpio10(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gpio_led = 0x10; /* GPIO4 */ + spec->default_polarity = 0; + } +} + static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2604,6 +2636,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { .chained = true, .chain_id = STAC_92HD83XXX_HP, }, + [STAC_HP_LED_GPIO10] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_hp_led_gpio10, + .chained = true, + .chain_id = STAC_92HD83XXX_HP, + }, [STAC_92HD83XXX_HEADSET_JACK] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd83xxx_fixup_headset_jack, @@ -2682,6 +2720,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1888, "HP Envy Spectre", STAC_HP_ENVY_BASS), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1899, + "HP Folio 13", STAC_HP_LED_GPIO10), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18df, "HP Folio", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18F8, @@ -4462,7 +4502,7 @@ static void stac_setup_gpio(struct hda_codec *codec) if (spec->mic_mute_led_gpio) { spec->gpio_mask |= spec->mic_mute_led_gpio; spec->gpio_dir |= spec->mic_mute_led_gpio; - spec->mic_mute_led_on = true; + spec->mic_enabled = 0; spec->gpio_data |= spec->mic_mute_led_gpio; spec->gen.cap_sync_hook = stac_capture_led_hook; diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 5799fbc24c28..8fe3b8c18ed4 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -39,6 +39,7 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) } static void update_tpacpi_micmute_led(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { if (!ucontrol || !led_set_func) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8cbb75..4544d8eb1452 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -11,7 +11,7 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio Codec Add-On Card support" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S if !BF60x select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_SSM2602 @@ -21,10 +21,9 @@ config SND_BF5XX_SOC_SSM2602 config SND_SOC_BFIN_EVAL_ADAU1701 tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && I2C select SND_BF5XX_SOC_I2S select SND_SOC_ADAU1701 - select I2C help Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ board connected to one of the Blackfin evaluation boards like the @@ -45,7 +44,7 @@ config SND_SOC_BFIN_EVAL_ADAU1373 config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_ADAV80X help @@ -58,7 +57,7 @@ config SND_SOC_BFIN_EVAL_ADAV80X config SND_BF5XX_SOC_AD1836 tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SPI_MASTER select SND_BF5XX_SOC_I2S select SND_SOC_AD1836 help @@ -66,7 +65,7 @@ config SND_BF5XX_SOC_AD1836 config SND_BF5XX_SOC_AD193X tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S select SND_SOC_AD193X help diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 75d0ad5d2dcb..647a72cda005 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1328,6 +1328,9 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; codec->control_data = pm860x->regmap; + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret) + return ret; for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 7257a8885f42..34d965a4a040 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -57,8 +57,8 @@ static const u16 ad1980_reg[] = { static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; -static const struct soc_enum ad1980_cap_src = - SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel); +static SOC_ENUM_DOUBLE_DECL(ad1980_cap_src, + AC97_REC_SEL, 8, 0, ad1980_rec_sel); static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = { SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b6569910..f4d965ebc29e 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1268,11 +1268,23 @@ static struct snd_soc_dai_driver da732x_dai[] = { }, }; +static bool da732x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA732X_REG_HPL_DAC_OFF_CNTL: + case DA732X_REG_HPR_DAC_OFF_CNTL: + return true; + default: + return false; + } +} + static const struct regmap_config da732x_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = DA732X_MAX_REG, + .volatile_reg = da732x_volatile, .reg_defaults = da732x_reg_cache, .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), .cache_type = REGCACHE_RBTREE, diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 52b79a487ac7..422812613a28 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1523,8 +1523,15 @@ static int da9055_remove(struct i2c_client *client) return 0; } +/* + * DO NOT change the device Ids. The naming is intentionally specific as both + * the CODEC and PMIC parts of this chip are instantiated separately as I2C + * devices (both have configurable I2C addresses, and are to all intents and + * purposes separate). As a result there are specific DA9055 Ids for CODEC + * and PMIC, which must be different to operate together. + */ static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, + { "da9055-codec", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); @@ -1532,7 +1539,7 @@ MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, }, .probe = da9055_i2c_probe, diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5839048ec467..cb736ddc446d 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -140,13 +140,17 @@ static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"}; static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"}; static const struct soc_enum isabelle_rx1_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts), - SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, + ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, + ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts), }; static const struct soc_enum isabelle_rx2_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts), - SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, + ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, + ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts), }; /* Headset DAC playback switches */ @@ -161,13 +165,17 @@ static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"}; static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"}; static const struct soc_enum isabelle_atx_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts), - SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, + ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, + ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts), }; static const struct soc_enum isabelle_vtx_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts), - SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, + ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, + ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts), }; static const struct snd_kcontrol_new atx_mux_controls = @@ -183,17 +191,13 @@ static const char *isabelle_amic1_texts[] = { /* Left analog microphone selection */ static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"}; -static const struct soc_enum isabelle_amic1_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5, - ARRAY_SIZE(isabelle_amic1_texts), - isabelle_amic1_texts), -}; +static SOC_ENUM_SINGLE_DECL(isabelle_amic1_enum, + ISABELLE_AMIC_CFG_REG, 5, + isabelle_amic1_texts); -static const struct soc_enum isabelle_amic2_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4, - ARRAY_SIZE(isabelle_amic2_texts), - isabelle_amic2_texts), -}; +static SOC_ENUM_SINGLE_DECL(isabelle_amic2_enum, + ISABELLE_AMIC_CFG_REG, 4, + isabelle_amic2_texts); static const struct snd_kcontrol_new amic1_control = SOC_DAPM_ENUM("Route", isabelle_amic1_enum); @@ -206,16 +210,20 @@ static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"}; static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"}; static const struct soc_enum isabelle_st_audio_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_audio_texts), isabelle_st_audio_texts), - SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_audio_texts), isabelle_st_audio_texts), }; static const struct soc_enum isabelle_st_voice_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_voice_texts), isabelle_st_voice_texts), - SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_voice_texts), isabelle_st_voice_texts), }; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1686ade2a429..f363de19be07 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -336,6 +336,7 @@ static bool max98090_readable_register(struct device *dev, unsigned int reg) case M98090_REG_RECORD_TDM_SLOT: case M98090_REG_SAMPLE_RATE: case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + case M98090_REG_REVISION_ID: return true; default: return false; @@ -1791,16 +1792,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regcache_sync(max98090->regmap); - - if (ret != 0) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } - } - if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { /* * Set to normal bias level. @@ -1814,6 +1805,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: /* Set internal pull-up to lowest power mode */ snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index fe4a5c2d4845..1a1e1150237d 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2086,6 +2086,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, + { "10EC5640", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 52e7cb08434b..fa2b8e07f420 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -210,7 +210,7 @@ out: static int si476x_codec_probe(struct snd_soc_codec *codec) { codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - return 0; + return snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); } static struct snd_soc_dai_ops si476x_dai_ops = { diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 06edb396e733..2735361a4c3c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -187,42 +187,42 @@ static const unsigned int sta32x_limiter_drc_release_tlv[] = { 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), }; -static const struct soc_enum sta32x_drc_ac_enum = - SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, - 2, sta32x_drc_ac); -static const struct soc_enum sta32x_auto_eq_enum = - SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, - 3, sta32x_auto_eq_mode); -static const struct soc_enum sta32x_auto_gc_enum = - SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, - 4, sta32x_auto_gc_mode); -static const struct soc_enum sta32x_auto_xo_enum = - SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, - 16, sta32x_auto_xo_mode); -static const struct soc_enum sta32x_preset_eq_enum = - SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, - 32, sta32x_preset_eq_mode); -static const struct soc_enum sta32x_limiter_ch1_enum = - SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter_ch2_enum = - SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter_ch3_enum = - SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter1_attack_rate_enum = - SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, - 16, sta32x_limiter_attack_rate); -static const struct soc_enum sta32x_limiter2_attack_rate_enum = - SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, - 16, sta32x_limiter_attack_rate); -static const struct soc_enum sta32x_limiter1_release_rate_enum = - SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, - 16, sta32x_limiter_release_rate); -static const struct soc_enum sta32x_limiter2_release_rate_enum = - SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, - 16, sta32x_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_drc_ac_enum, + STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + sta32x_drc_ac); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_eq_enum, + STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + sta32x_auto_eq_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_gc_enum, + STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + sta32x_auto_gc_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_xo_enum, + STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + sta32x_auto_xo_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_preset_eq_enum, + STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + sta32x_preset_eq_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch1_enum, + STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch2_enum, + STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch3_enum, + STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_attack_rate_enum, + STA32X_L1AR, STA32X_LxA_SHIFT, + sta32x_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_attack_rate_enum, + STA32X_L2AR, STA32X_LxA_SHIFT, + sta32x_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_release_rate_enum, + STA32X_L1AR, STA32X_LxR_SHIFT, + sta32x_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, + STA32X_L2AR, STA32X_LxR_SHIFT, + sta32x_limiter_release_rate); /* byte array controls for setting biquad, mixer, scaling coefficients; * for biquads all five coefficients need to be set in one go, @@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) static int sta32x_cache_sync(struct snd_soc_codec *codec) { - struct sta32x_priv *sta32x = codec->control_data; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int mute; int rc; @@ -434,7 +434,7 @@ SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), -SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter2_release_rate_enum), /* depending on mode, the attack/release thresholds have * two different enum definitions; provide both diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 48dc7d2fee36..6d684d934f4d 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -117,19 +117,23 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, static const char *wm8400_digital_sidetone[] = {"None", "Left ADC", "Right ADC", "Reserved"}; -static const struct soc_enum wm8400_left_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, - WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8400_left_digital_sidetone_enum, + WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACL_SHIFT, + wm8400_digital_sidetone); -static const struct soc_enum wm8400_right_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, - WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8400_right_digital_sidetone_enum, + WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACR_SHIFT, + wm8400_digital_sidetone); static const char *wm8400_adcmode[] = {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; -static const struct soc_enum wm8400_right_adcmode_enum = -SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode); +static SOC_ENUM_SINGLE_DECL(wm8400_right_adcmode_enum, + WM8400_ADC_CTRL, + WM8400_ADC_HPF_CUT_SHIFT, + wm8400_adcmode); static const struct snd_kcontrol_new wm8400_snd_controls[] = { /* INMIXL */ @@ -422,9 +426,10 @@ SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT, static const char *wm8400_ainlmux[] = {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; -static const struct soc_enum wm8400_ainlmux_enum = -SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT, - ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux); +static SOC_ENUM_SINGLE_DECL(wm8400_ainlmux_enum, + WM8400_INPUT_MIXER1, + WM8400_AINLMODE_SHIFT, + wm8400_ainlmux); static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); @@ -435,9 +440,10 @@ SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); static const char *wm8400_ainrmux[] = {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; -static const struct soc_enum wm8400_ainrmux_enum = -SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT, - ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux); +static SOC_ENUM_SINGLE_DECL(wm8400_ainrmux_enum, + WM8400_INPUT_MIXER1, + WM8400_AINRMODE_SHIFT, + wm8400_ainrmux); static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum); diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 89a18d82f303..5bce21013485 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -196,8 +196,8 @@ static const char *ain_text[] = { "AIN5", "AIN6", "AIN7", "AIN8" }; -static const struct soc_enum ain_enum = - SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text); +static SOC_ENUM_DOUBLE_DECL(ain_enum, + WM8770_ADCMUX, 0, 4, ain_text); static const struct snd_kcontrol_new ain_mux = SOC_DAPM_ENUM("Capture Mux", ain_enum); diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e98bc7038a08..43c2201cb901 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -304,53 +304,53 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1); static const char *mic_bias_level_txt[] = { "0.9*AVDD", "0.65*AVDD" }; -static const struct soc_enum mic_bias_level = -SOC_ENUM_SINGLE(WM8900_REG_INCTL, 8, 2, mic_bias_level_txt); +static SOC_ENUM_SINGLE_DECL(mic_bias_level, + WM8900_REG_INCTL, 8, mic_bias_level_txt); static const char *dac_mute_rate_txt[] = { "Fast", "Slow" }; -static const struct soc_enum dac_mute_rate = -SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 7, 2, dac_mute_rate_txt); +static SOC_ENUM_SINGLE_DECL(dac_mute_rate, + WM8900_REG_DACCTRL, 7, dac_mute_rate_txt); static const char *dac_deemphasis_txt[] = { "Disabled", "32kHz", "44.1kHz", "48kHz" }; -static const struct soc_enum dac_deemphasis = -SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 4, 4, dac_deemphasis_txt); +static SOC_ENUM_SINGLE_DECL(dac_deemphasis, + WM8900_REG_DACCTRL, 4, dac_deemphasis_txt); static const char *adc_hpf_cut_txt[] = { "Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3" }; -static const struct soc_enum adc_hpf_cut = -SOC_ENUM_SINGLE(WM8900_REG_ADCCTRL, 5, 4, adc_hpf_cut_txt); +static SOC_ENUM_SINGLE_DECL(adc_hpf_cut, + WM8900_REG_ADCCTRL, 5, adc_hpf_cut_txt); static const char *lr_txt[] = { "Left", "Right" }; -static const struct soc_enum aifl_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 15, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(aifl_src, + WM8900_REG_AUDIO1, 15, lr_txt); -static const struct soc_enum aifr_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 14, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(aifr_src, + WM8900_REG_AUDIO1, 14, lr_txt); -static const struct soc_enum dacl_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 15, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(dacl_src, + WM8900_REG_AUDIO2, 15, lr_txt); -static const struct soc_enum dacr_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 14, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(dacr_src, + WM8900_REG_AUDIO2, 14, lr_txt); static const char *sidetone_txt[] = { "Disabled", "Left ADC", "Right ADC" }; -static const struct soc_enum dacl_sidetone = -SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 2, 3, sidetone_txt); +static SOC_ENUM_SINGLE_DECL(dacl_sidetone, + WM8900_REG_SIDETONE, 2, sidetone_txt); -static const struct soc_enum dacr_sidetone = -SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 0, 3, sidetone_txt); +static SOC_ENUM_SINGLE_DECL(dacr_sidetone, + WM8900_REG_SIDETONE, 0, sidetone_txt); static const struct snd_kcontrol_new wm8900_snd_controls[] = { SOC_ENUM("Mic Bias Level", mic_bias_level), @@ -496,8 +496,8 @@ SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0), static const char *wm8900_lp_mux[] = { "Disabled", "Enabled" }; -static const struct soc_enum wm8900_lineout2_lp_mux = -SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm8900_lp_mux); +static SOC_ENUM_SINGLE_DECL(wm8900_lineout2_lp_mux, + WM8900_REG_LOUTMIXCTL1, 1, wm8900_lp_mux); static const struct snd_kcontrol_new wm8900_lineout2_lp = SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux); diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 19743779bf4d..7ac2e511403c 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -153,7 +153,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, data32 &= 0xffffff; - wm8994_bulk_write(codec->control_data, + wm8994_bulk_write(wm8994->wm8994, data32 & 0xffffff, block_len / 2, (void *)(data + 8)); diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a0f3ef..2ee23a39622c 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1562,7 +1562,6 @@ static int wm8993_remove(struct snd_soc_codec *codec) struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); return 0; } diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 70ff3772079f..5e3bc3c6801a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -399,6 +399,7 @@ static struct platform_driver davinci_evm_driver = { .driver = { .name = "davinci_evm", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = of_match_ptr(davinci_evm_dt_ids), }, }; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b7858bfa0295..670afa29e30d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -263,7 +263,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + pm_runtime_get_sync(mcasp->dev); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: @@ -317,7 +319,8 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; default: - return -EINVAL; + ret = -EINVAL; + goto out; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -354,10 +357,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; default: - return -EINVAL; + ret = -EINVAL; + break; } - - return 0; +out: + pm_runtime_put_sync(mcasp->dev); + return ret; } static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) @@ -448,7 +453,7 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, return 0; } -static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, +static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int channels) { int i; @@ -524,12 +529,18 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, return 0; } -static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) +static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) { int i, active_slots; u32 mask = 0; u32 busel = 0; + if ((mcasp->tdm_slots < 2) || (mcasp->tdm_slots > 32)) { + dev_err(mcasp->dev, "tdm slot %d not supported\n", + mcasp->tdm_slots); + return -EINVAL; + } + active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots; for (i = 0; i < active_slots; i++) mask |= (1 << i); @@ -539,35 +550,21 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) if (!mcasp->dat_port) busel = TXSEL; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* bit stream is MSB first with no delay */ - /* DSP_B mode */ - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); - - if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); - else - printk(KERN_ERR "playback tdm slot %d not supported\n", - mcasp->tdm_slots); - } else { - /* bit stream is MSB first with no delay */ - /* DSP_B mode */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); - - if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); - else - printk(KERN_ERR "capture tdm slot %d not supported\n", - mcasp->tdm_slots); - } + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); + + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); + + return 0; } /* S/PDIF */ -static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) { /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ @@ -589,6 +586,8 @@ static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + + return 0; } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, @@ -605,13 +604,14 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, u8 slots = mcasp->tdm_slots; u8 active_serializers; int channels; + int ret; struct snd_interval *pcm_channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); channels = pcm_channels->min; active_serializers = (channels + slots - 1) / slots; - if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL) + if (mcasp_common_hw_param(mcasp, substream->stream, channels) == -EINVAL) return -EINVAL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) fifo_level = mcasp->txnumevt * active_serializers; @@ -619,9 +619,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, fifo_level = mcasp->rxnumevt * active_serializers; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - davinci_hw_dit_param(mcasp); + ret = mcasp_dit_hw_param(mcasp); else - davinci_hw_param(mcasp, substream->stream); + ret = mcasp_i2s_hw_param(mcasp, substream->stream); + + if (ret) + return ret; switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: @@ -678,19 +681,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = pm_runtime_get_sync(mcasp->dev); - if (IS_ERR_VALUE(ret)) - dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n"); davinci_mcasp_start(mcasp, substream->stream); break; - case SNDRV_PCM_TRIGGER_SUSPEND: - davinci_mcasp_stop(mcasp, substream->stream); - ret = pm_runtime_put_sync(mcasp->dev); - if (IS_ERR_VALUE(ret)) - dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n"); - break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(mcasp, substream->stream); diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0c72ed261e7..c84026c99134 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -326,7 +326,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); @@ -334,7 +334,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 9c9f957fcae1..75e14033e8d8 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -322,7 +322,7 @@ #define ESAI_xSMB_xS_SHIFT 0 #define ESAI_xSMB_xS_WIDTH 16 #define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) -#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK) /* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ #define ESAI_PRRC_PDC_SHIFT 0 diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 79cee782dbbf..a2fd7321b5a9 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,7 +160,6 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f2beae78969f..1cb22dd034eb 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -33,8 +33,7 @@ struct imx_sgtl5000_data { static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct imx_sgtl5000_data *data = container_of(rtd->card, - struct imx_sgtl5000_data, card); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card); struct device *dev = rtd->card->dev; int ret; @@ -159,13 +158,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -184,7 +185,8 @@ fail: static int imx_sgtl5000_remove(struct platform_device *pdev) { - struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card); clk_put(data->codec_clk); diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc391de..3a3d17ce6ba4 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; @@ -137,7 +137,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; @@ -264,13 +264,15 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.late_probe = imx_wm8962_late_probe; data->card.set_bias_level = imx_wm8962_set_bias_level; + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -289,7 +291,8 @@ fail: static int imx_wm8962_remove(struct platform_device *pdev) { - struct imx_wm8962_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 480a39ce02bc..fd4d9c809e50 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -309,7 +309,9 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) + if (!of_have_populated_dt() || + (!of_machine_is_compatible("nokia,n810") && + !of_machine_is_compatible("nokia,n810-wimax"))) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 454f41cfc828..350757400391 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -59,7 +59,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 select SND_SOC_WM8750 select SND_S3C2412_SOC_I2S help - Sat Y if you want to add support for SoC audio on the Jive. + Say Y if you want to add support for SoC audio on the Jive. config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" @@ -145,11 +145,11 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help - Sat Y if you want to add support for SoC audio on the SMDK. + Say Y if you want to add support for SoC audio on the SMDK. config SND_SOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 375dc6dfba4e..bfed3e4c45ff 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -96,8 +96,7 @@ int snd_soc_cache_exit(struct snd_soc_codec *codec) { dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n", codec->name); - if (!codec->reg_cache) - return 0; + kfree(codec->reg_cache); codec->reg_cache = NULL; return 0; @@ -117,8 +116,9 @@ int snd_soc_cache_read(struct snd_soc_codec *codec, return -EINVAL; mutex_lock(&codec->cache_rw_mutex); - *value = snd_soc_get_cache_val(codec->reg_cache, reg, - codec->driver->reg_word_size); + if (!ZERO_OR_NULL_PTR(codec->reg_cache)) + *value = snd_soc_get_cache_val(codec->reg_cache, reg, + codec->driver->reg_word_size); mutex_unlock(&codec->cache_rw_mutex); return 0; @@ -136,8 +136,9 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { mutex_lock(&codec->cache_rw_mutex); - snd_soc_set_cache_val(codec->reg_cache, reg, value, - codec->driver->reg_word_size); + if (!ZERO_OR_NULL_PTR(codec->reg_cache)) + snd_soc_set_cache_val(codec->reg_cache, reg, value, + codec->driver->reg_word_size); mutex_unlock(&codec->cache_rw_mutex); return 0; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1fa7dda2e2bf..4ba0959a0d7b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -56,7 +56,6 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); -static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); static LIST_HEAD(component_list); @@ -370,18 +369,22 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf, { char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); ssize_t len, ret = 0; + struct snd_soc_component *component; struct snd_soc_dai *dai; if (!buf) return -ENOMEM; - list_for_each_entry(dai, &dai_list, list) { - len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", dai->name); - if (len >= 0) - ret += len; - if (ret > PAGE_SIZE) { - ret = PAGE_SIZE; - break; + list_for_each_entry(component, &component_list, list) { + list_for_each_entry(dai, &component->dai_list, list) { + len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", + dai->name); + if (len >= 0) + ret += len; + if (ret > PAGE_SIZE) { + ret = PAGE_SIZE; + break; + } } } @@ -855,6 +858,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_component *component; struct snd_soc_codec *codec; struct snd_soc_platform *platform; struct snd_soc_dai *codec_dai, *cpu_dai; @@ -863,18 +867,20 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); /* Find CPU DAI from registered DAIs*/ - list_for_each_entry(cpu_dai, &dai_list, list) { + list_for_each_entry(component, &component_list, list) { if (dai_link->cpu_of_node && - (cpu_dai->dev->of_node != dai_link->cpu_of_node)) + component->dev->of_node != dai_link->cpu_of_node) continue; if (dai_link->cpu_name && - strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name)) - continue; - if (dai_link->cpu_dai_name && - strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + strcmp(dev_name(component->dev), dai_link->cpu_name)) continue; + list_for_each_entry(cpu_dai, &component->dai_list, list) { + if (dai_link->cpu_dai_name && + strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; - rtd->cpu_dai = cpu_dai; + rtd->cpu_dai = cpu_dai; + } } if (!rtd->cpu_dai) { @@ -899,12 +905,10 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) * CODEC found, so find CODEC DAI from registered DAIs from * this CODEC */ - list_for_each_entry(codec_dai, &dai_list, list) { - if (codec->dev == codec_dai->dev && - !strcmp(codec_dai->name, - dai_link->codec_dai_name)) { - + list_for_each_entry(codec_dai, &codec->component.dai_list, list) { + if (!strcmp(codec_dai->name, dai_link->codec_dai_name)) { rtd->codec_dai = codec_dai; + break; } } @@ -1128,12 +1132,8 @@ static int soc_probe_codec(struct snd_soc_card *card, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &dai_list, list) { - if (dai->dev != codec->dev) - continue; - + list_for_each_entry(dai, &codec->component.dai_list, list) snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); - } codec->dapm.idle_bias_off = driver->idle_bias_off; @@ -1180,6 +1180,7 @@ static int soc_probe_platform(struct snd_soc_card *card, { int ret = 0; const struct snd_soc_platform_driver *driver = platform->driver; + struct snd_soc_component *component; struct snd_soc_dai *dai; platform->card = card; @@ -1195,11 +1196,11 @@ static int soc_probe_platform(struct snd_soc_card *card, driver->dapm_widgets, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &dai_list, list) { - if (dai->dev != platform->dev) + list_for_each_entry(component, &component_list, list) { + if (component->dev != platform->dev) continue; - - snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); + list_for_each_entry(dai, &component->dai_list, list) + snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); } platform->dapm.idle_bias_off = 1; @@ -2800,7 +2801,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; int err; - bool type_2r = 0; + bool type_2r = false; unsigned int val2 = 0; unsigned int val, val_mask; @@ -2821,7 +2822,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val |= val2 << rshift; } else { val2 = val2 << shift; - type_2r = 1; + type_2r = true; } } err = snd_soc_update_bits_locked(codec, reg, val_mask, val); @@ -3219,7 +3220,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct soc_bytes *params = (void *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); int ret, len; - unsigned int val; + unsigned int val, mask; void *data; if (!codec->using_regmap) @@ -3249,12 +3250,36 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ((u8 *)data)[0] |= val; break; case 2: - ((u16 *)data)[0] &= cpu_to_be16(~params->mask); - ((u16 *)data)[0] |= cpu_to_be16(val); + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, + &mask, &mask); + if (ret != 0) + goto out; + + ((u16 *)data)[0] &= mask; + + ret = regmap_parse_val(codec->control_data, + &val, &val); + if (ret != 0) + goto out; + + ((u16 *)data)[0] |= val; break; case 4: - ((u32 *)data)[0] &= cpu_to_be32(~params->mask); - ((u32 *)data)[0] |= cpu_to_be32(val); + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, + &mask, &mask); + if (ret != 0) + goto out; + + ((u32 *)data)[0] &= mask; + + ret = regmap_parse_val(codec->control_data, + &val, &val); + if (ret != 0) + goto out; + + ((u32 *)data)[0] |= val; break; default: ret = -EINVAL; @@ -3611,7 +3636,7 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, slots, slot_width); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); @@ -3867,100 +3892,42 @@ static inline char *fmt_multiple_name(struct device *dev, } /** - * snd_soc_register_dai - Register a DAI with the ASoC core + * snd_soc_unregister_dai - Unregister DAIs from the ASoC core * - * @component: The component the DAIs are registered for - * @dai_drv: DAI driver to use for the DAIs + * @component: The component for which the DAIs should be unregistered */ -static int snd_soc_register_dai(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv) +static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct device *dev = component->dev; - struct snd_soc_codec *codec; - struct snd_soc_dai *dai; - - dev_dbg(dev, "ASoC: dai register %s\n", dev_name(dev)); - - dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); - if (dai == NULL) - return -ENOMEM; + struct snd_soc_dai *dai, *_dai; - /* create DAI component name */ - dai->name = fmt_single_name(dev, &dai->id); - if (dai->name == NULL) { + list_for_each_entry_safe(dai, _dai, &component->dai_list, list) { + dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n", + dai->name); + list_del(&dai->list); + kfree(dai->name); kfree(dai); - return -ENOMEM; - } - - dai->component = component; - dai->dev = dev; - dai->driver = dai_drv; - dai->dapm.dev = dev; - if (!dai->driver->ops) - dai->driver->ops = &null_dai_ops; - - mutex_lock(&client_mutex); - - list_for_each_entry(codec, &codec_list, list) { - if (codec->dev == dev) { - dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", - dai->name, codec->name); - dai->codec = codec; - break; - } - } - - if (!dai->codec) - dai->dapm.idle_bias_off = 1; - - list_add(&dai->list, &dai_list); - - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); - - return 0; -} - -/** - * snd_soc_unregister_dai - Unregister a DAI from the ASoC core - * - * @dai: DAI to unregister - */ -static void snd_soc_unregister_dai(struct device *dev) -{ - struct snd_soc_dai *dai; - - list_for_each_entry(dai, &dai_list, list) { - if (dev == dai->dev) - goto found; } - return; - -found: - mutex_lock(&client_mutex); - list_del(&dai->list); - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Unregistered DAI '%s'\n", dai->name); - kfree(dai->name); - kfree(dai); } /** - * snd_soc_register_dais - Register multiple DAIs with the ASoC core + * snd_soc_register_dais - Register a DAI with the ASoC core * * @component: The component the DAIs are registered for + * @codec: The CODEC that the DAIs are registered for, NULL if the component is + * not a CODEC. * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs + * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the + * parent's name. */ static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, size_t count) + struct snd_soc_codec *codec, struct snd_soc_dai_driver *dai_drv, + size_t count, bool legacy_dai_naming) { struct device *dev = component->dev; - struct snd_soc_codec *codec; struct snd_soc_dai *dai; - int i, ret = 0; + unsigned int i; + int ret; dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count); @@ -3972,71 +3939,54 @@ static int snd_soc_register_dais(struct snd_soc_component *component, goto err; } - /* create DAI component name */ - dai->name = fmt_multiple_name(dev, &dai_drv[i]); + /* + * Back in the old days when we still had component-less DAIs, + * instead of having a static name, component-less DAIs would + * inherit the name of the parent device so it is possible to + * register multiple instances of the DAI. We still need to keep + * the same naming style even though those DAIs are not + * component-less anymore. + */ + if (count == 1 && legacy_dai_naming) { + dai->name = fmt_single_name(dev, &dai->id); + } else { + dai->name = fmt_multiple_name(dev, &dai_drv[i]); + if (dai_drv[i].id) + dai->id = dai_drv[i].id; + else + dai->id = i; + } if (dai->name == NULL) { kfree(dai); - ret = -EINVAL; + ret = -ENOMEM; goto err; } dai->component = component; + dai->codec = codec; dai->dev = dev; dai->driver = &dai_drv[i]; - if (dai->driver->id) - dai->id = dai->driver->id; - else - dai->id = i; dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; - mutex_lock(&client_mutex); - - list_for_each_entry(codec, &codec_list, list) { - if (codec->dev == dev) { - dev_dbg(dev, - "ASoC: Mapped DAI %s to CODEC %s\n", - dai->name, codec->name); - dai->codec = codec; - break; - } - } - if (!dai->codec) dai->dapm.idle_bias_off = 1; - list_add(&dai->list, &dai_list); - - mutex_unlock(&client_mutex); + list_add(&dai->list, &component->dai_list); - dev_dbg(dai->dev, "ASoC: Registered DAI '%s'\n", dai->name); + dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); } return 0; err: - for (i--; i >= 0; i--) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(component); return ret; } /** - * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core - * - * @dai: Array of DAIs to unregister - * @count: Number of DAIs - */ -static void snd_soc_unregister_dais(struct device *dev, size_t count) -{ - int i; - - for (i = 0; i < count; i++) - snd_soc_unregister_dai(dev); -} - -/** * snd_soc_register_component - Register a component with the ASoC core * */ @@ -4044,6 +3994,7 @@ static int __snd_soc_register_component(struct device *dev, struct snd_soc_component *cmpnt, const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_codec *codec, struct snd_soc_dai_driver *dai_drv, int num_dai, bool allow_single_dai) { @@ -4066,20 +4017,10 @@ __snd_soc_register_component(struct device *dev, cmpnt->driver = cmpnt_drv; cmpnt->dai_drv = dai_drv; cmpnt->num_dai = num_dai; + INIT_LIST_HEAD(&cmpnt->dai_list); - /* - * snd_soc_register_dai() uses fmt_single_name(), and - * snd_soc_register_dais() uses fmt_multiple_name() - * for dai->name which is used for name based matching - * - * this function is used from cpu/codec. - * allow_single_dai flag can ignore "codec" driver reworking - * since it had been used snd_soc_register_dais(), - */ - if ((1 == num_dai) && allow_single_dai) - ret = snd_soc_register_dai(cmpnt, dai_drv); - else - ret = snd_soc_register_dais(cmpnt, dai_drv, num_dai); + ret = snd_soc_register_dais(cmpnt, codec, dai_drv, num_dai, + allow_single_dai); if (ret < 0) { dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); goto error_component_name; @@ -4114,7 +4055,7 @@ int snd_soc_register_component(struct device *dev, cmpnt->ignore_pmdown_time = true; - return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, + return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, NULL, dai_drv, num_dai, true); } EXPORT_SYMBOL_GPL(snd_soc_register_component); @@ -4134,7 +4075,7 @@ void snd_soc_unregister_component(struct device *dev) return; found: - snd_soc_unregister_dais(dev, cmpnt->num_dai); + snd_soc_unregister_dais(cmpnt); mutex_lock(&client_mutex); list_del(&cmpnt->list); @@ -4335,7 +4276,7 @@ int snd_soc_register_codec(struct device *dev, /* register component */ ret = __snd_soc_register_component(dev, &codec->component, &codec_drv->component_driver, - dai_drv, num_dai, false); + codec, dai_drv, num_dai, false); if (ret < 0) { dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret); goto fail_codec_name; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3651a37bb2c7..c8a780d0d057 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -111,6 +111,12 @@ static int dapm_down_seq[] = { [snd_soc_dapm_post] = 14, }; +static void dapm_assert_locked(struct snd_soc_dapm_context *dapm) +{ + if (dapm->card && dapm->card->instantiated) + lockdep_assert_held(&dapm->card->dapm_mutex); +} + static void pop_wait(u32 pop_time) { if (pop_time) @@ -142,15 +148,16 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) +static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { + dapm_assert_locked(w->dapm); + if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", w->name, reason); list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); } } -EXPORT_SYMBOL_GPL(dapm_mark_dirty); void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) { @@ -357,6 +364,8 @@ static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; + lockdep_assert_held(&card->dapm_mutex); + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); list_for_each_entry(w, &card->widgets, list) { @@ -1751,6 +1760,8 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) ASYNC_DOMAIN_EXCLUSIVE(async_domain); enum snd_soc_bias_level bias; + lockdep_assert_held(&card->dapm_mutex); + trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { @@ -1825,10 +1836,14 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) trace_snd_soc_dapm_walk_done(card); - /* Run all the bias changes in parallel */ - list_for_each_entry(d, &card->dapm_list, list) - async_schedule_domain(dapm_pre_sequence_async, d, - &async_domain); + /* Run card bias changes at first */ + dapm_pre_sequence_async(&card->dapm, 0); + /* Run other bias changes in parallel */ + list_for_each_entry(d, &card->dapm_list, list) { + if (d != &card->dapm) + async_schedule_domain(dapm_pre_sequence_async, d, + &async_domain); + } async_synchronize_full_domain(&async_domain); list_for_each_entry(w, &down_list, power_list) { @@ -1848,10 +1863,14 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_seq_run(card, &up_list, event, true); /* Run all the bias changes in parallel */ - list_for_each_entry(d, &card->dapm_list, list) - async_schedule_domain(dapm_post_sequence_async, d, - &async_domain); + list_for_each_entry(d, &card->dapm_list, list) { + if (d != &card->dapm) + async_schedule_domain(dapm_post_sequence_async, d, + &async_domain); + } async_synchronize_full_domain(&async_domain); + /* Run card bias changes at last */ + dapm_post_sequence_async(&card->dapm, 0); /* do we need to notify any clients that DAPM event is complete */ list_for_each_entry(d, &card->dapm_list, list) { @@ -2038,6 +2057,8 @@ static int soc_dapm_mux_update_power(struct snd_soc_card *card, struct snd_soc_dapm_path *path; int found = 0; + lockdep_assert_held(&card->dapm_mutex); + /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { if (!path->name || !e->texts[mux]) @@ -2088,6 +2109,8 @@ static int soc_dapm_mixer_update_power(struct snd_soc_card *card, struct snd_soc_dapm_path *path; int found = 0; + lockdep_assert_held(&card->dapm_mutex); + /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { found = 1; @@ -2253,6 +2276,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + dapm_assert_locked(dapm); + if (!w) { dev_err(dapm->dev, "ASoC: DAPM unknown pin %s\n", pin); return -EINVAL; @@ -3899,7 +3924,7 @@ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); -static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) +static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm) { struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; @@ -3939,14 +3964,21 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) */ void snd_soc_dapm_shutdown(struct snd_soc_card *card) { - struct snd_soc_codec *codec; + struct snd_soc_dapm_context *dapm; - list_for_each_entry(codec, &card->codec_dev_list, card_list) { - soc_dapm_shutdown_codec(&codec->dapm); - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) - snd_soc_dapm_set_bias_level(&codec->dapm, - SND_SOC_BIAS_OFF); + list_for_each_entry(dapm, &card->dapm_list, list) { + if (dapm != &card->dapm) { + soc_dapm_shutdown_dapm(dapm); + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_OFF); + } } + + soc_dapm_shutdown_dapm(&card->dapm); + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(&card->dapm, + SND_SOC_BIAS_OFF); } /* Module information */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2cedf09f6d96..330eaf007d89 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2050,6 +2050,7 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card) paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list); if (paths < 0) { + dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "playback"); mutex_unlock(&card->mutex); @@ -2079,6 +2080,7 @@ capture: paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list); if (paths < 0) { + dpcm_path_put(&list); dev_warn(fe->dev, "ASoC: %s no valid %s path\n", fe->dai_link->name, "capture"); mutex_unlock(&card->mutex); @@ -2143,6 +2145,7 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { + dpcm_path_put(&list); dev_dbg(fe->dev, "ASoC: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); } diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index fe99f461aff0..19cca043e6e4 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -213,10 +213,7 @@ static int spdif_digital_mute(struct snd_soc_dai *dai, int mute) static int spdif_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_card *card = codec->card; - struct snd_soc_pcm_runtime *rtd = card->rtd; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); ucontrol->value.integer.value[0] = host->saved_params.mute; @@ -226,10 +223,7 @@ static int spdif_mute_get(struct snd_kcontrol *kcontrol, static int spdif_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_card *card = codec->card; - struct snd_soc_pcm_runtime *rtd = card->rtd; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); if (host->saved_params.mute == ucontrol->value.integer.value[0]) diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index e0305a148568..9edd68db9f48 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -183,14 +183,16 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); drvdata->base = devm_ioremap_resource(&pdev->dev, r); if (IS_ERR(drvdata->base)) return PTR_ERR(drvdata->base); - drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); - if (!drvdata) - return -ENOMEM; platform_set_drvdata(pdev, drvdata); drvdata->physbase = r->start; if (sizeof(drvdata->physbase) > sizeof(r->start) && diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b83f75..e05a86b7c0da 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -14,6 +14,7 @@ config SND_USB_AUDIO select SND_HWDEP select SND_RAWMIDI select SND_PCM + select BITREVERSE help Say Y here to include support for USB audio and USB MIDI devices. diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 44b0ba4feab3..1bed780e21d9 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -883,6 +883,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } break; + case USB_ID(0x046d, 0x0807): /* Logitech Webcam C500 */ case USB_ID(0x046d, 0x0808): case USB_ID(0x046d, 0x0809): case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 32af6b741ef5..d1d72ff50347 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -328,6 +328,11 @@ static struct usbmix_name_map gamecom780_map[] = { {} }; +static const struct usbmix_name_map kef_x300a_map[] = { + { 10, NULL }, /* firmware locks up (?) when we try to access this FU */ + { 0 } +}; + /* * Control map entries */ @@ -419,6 +424,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x200c, 0x1018), .map = ebox44_map, }, + { + .id = USB_ID(0x27ac, 0x1000), + .map = kef_x300a_map, + }, { 0 } /* terminator */ }; |