diff options
Diffstat (limited to 'sound')
31 files changed, 254 insertions, 137 deletions
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 552b97afbca5..61ab640e195f 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -113,6 +113,7 @@ MODULE_ALIAS("sound-layout-100"); MODULE_ALIAS("aoa-device-id-14"); MODULE_ALIAS("aoa-device-id-22"); MODULE_ALIAS("aoa-device-id-35"); +MODULE_ALIAS("aoa-device-id-44"); /* onyx with all but microphone connected */ static struct codec_connection onyx_connections_nomic[] = { @@ -361,6 +362,13 @@ static struct layout layouts[] = { .connections = tas_connections_nolineout, }, }, + /* PowerBook6,5 */ + { .device_id = 44, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, /* PowerBook6,7 */ { .layout_id = 80, .codecs[0] = { diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 010658335881..15e76131b501 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -200,7 +200,8 @@ static int i2sbus_add_dev(struct macio_dev *macio, * We probably cannot handle all device-id machines, * so restrict to those we do handle for now. */ - if (id && (*id == 22 || *id == 14 || *id == 35)) { + if (id && (*id == 22 || *id == 14 || *id == 35 || + *id == 44)) { snprintf(dev->sound.modalias, 32, "aoa-device-id-%d", *id); ok = 1; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ccfa383f1fda..f92818155958 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1649,6 +1649,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) } if (!snd_pcm_stream_linked(substream)) { substream->group = group; + group = NULL; spin_lock_init(&substream->group->lock); INIT_LIST_HEAD(&substream->group->substreams); list_add_tail(&substream->link_list, &substream->group->substreams); @@ -1663,8 +1664,7 @@ static int snd_pcm_link(struct snd_pcm_substream *substream, int fd) _nolock: snd_card_unref(substream1->pcm->card); fput_light(file, fput_needed); - if (res < 0) - kfree(group); + kfree(group); return res; } diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig index 51c4ba95a32d..1a9640254433 100644 --- a/sound/oss/Kconfig +++ b/sound/oss/Kconfig @@ -250,7 +250,7 @@ config MSND_FIFOSIZE menuconfig SOUND_OSS tristate "OSS sound modules" depends on ISA_DMA_API && VIRT_TO_BUS - depends on !ISA_DMA_SUPPORT_BROKEN + depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN help OSS is the Open Sound System suite of sound card drivers. They make sound programming easier since they provide a common API. Say Y or diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac079f93c535..4b1524a861f3 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -606,6 +606,10 @@ static bool is_active_nid(struct hda_codec *codec, hda_nid_t nid, return false; } +/* check whether the NID is referred by any active paths */ +#define is_active_nid_for_any(codec, nid) \ + is_active_nid(codec, nid, HDA_OUTPUT, 0) + /* get the default amp value for the target state */ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps, bool enable) @@ -759,7 +763,8 @@ static void path_power_down_sync(struct hda_codec *codec, struct nid_path *path) for (i = 0; i < path->depth; i++) { hda_nid_t nid = path->path[i]; - if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3)) { + if (!snd_hda_check_power_state(codec, nid, AC_PWRST_D3) && + !is_active_nid_for_any(codec, nid)) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); @@ -783,6 +788,8 @@ static void set_pin_eapd(struct hda_codec *codec, hda_nid_t pin, bool enable) return; if (codec->inv_eapd) enable = !enable; + if (spec->keep_eapd_on && !enable) + return; snd_hda_codec_update_cache(codec, pin, 0, AC_VERB_SET_EAPD_BTLENABLE, enable ? 0x02 : 0x00); @@ -1933,17 +1940,7 @@ static int create_speaker_out_ctls(struct hda_codec *codec) * independent HP controls */ -/* update HP auto-mute state too */ -static void update_hp_automute_hook(struct hda_codec *codec) -{ - struct hda_gen_spec *spec = codec->spec; - - if (spec->hp_automute_hook) - spec->hp_automute_hook(codec, NULL); - else - snd_hda_gen_hp_automute(codec, NULL); -} - +static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack); static int indep_hp_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2004,7 +2001,7 @@ static int indep_hp_put(struct snd_kcontrol *kcontrol, else *dacp = spec->alt_dac_nid; - update_hp_automute_hook(codec); + call_hp_automute(codec, NULL); ret = 1; } unlock: @@ -2300,7 +2297,7 @@ static void update_hp_mic(struct hda_codec *codec, int adc_mux, bool force) else val = PIN_HP; set_pin_target(codec, pin, val, true); - update_hp_automute_hook(codec); + call_hp_automute(codec, NULL); } } @@ -2709,7 +2706,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol, val = snd_hda_get_default_vref(codec, nid); } snd_hda_set_pin_ctl_cache(codec, nid, val); - update_hp_automute_hook(codec); + call_hp_automute(codec, NULL); return 1; } @@ -3854,20 +3851,42 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja } EXPORT_SYMBOL_HDA(snd_hda_gen_mic_autoswitch); -/* update jack retasking */ -static void update_automute_all(struct hda_codec *codec) +/* call appropriate hooks */ +static void call_hp_automute(struct hda_codec *codec, struct hda_jack_tbl *jack) { struct hda_gen_spec *spec = codec->spec; + if (spec->hp_automute_hook) + spec->hp_automute_hook(codec, jack); + else + snd_hda_gen_hp_automute(codec, jack); +} - update_hp_automute_hook(codec); +static void call_line_automute(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct hda_gen_spec *spec = codec->spec; if (spec->line_automute_hook) - spec->line_automute_hook(codec, NULL); + spec->line_automute_hook(codec, jack); else - snd_hda_gen_line_automute(codec, NULL); + snd_hda_gen_line_automute(codec, jack); +} + +static void call_mic_autoswitch(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct hda_gen_spec *spec = codec->spec; if (spec->mic_autoswitch_hook) - spec->mic_autoswitch_hook(codec, NULL); + spec->mic_autoswitch_hook(codec, jack); else - snd_hda_gen_mic_autoswitch(codec, NULL); + snd_hda_gen_mic_autoswitch(codec, jack); +} + +/* update jack retasking */ +static void update_automute_all(struct hda_codec *codec) +{ + call_hp_automute(codec, NULL); + call_line_automute(codec, NULL); + call_mic_autoswitch(codec, NULL); } /* @@ -4004,9 +4023,7 @@ static int check_auto_mute_availability(struct hda_codec *codec) snd_printdd("hda-codec: Enable HP auto-muting on NID 0x%x\n", nid); snd_hda_jack_detect_enable_callback(codec, nid, HDA_GEN_HP_EVENT, - spec->hp_automute_hook ? - spec->hp_automute_hook : - snd_hda_gen_hp_automute); + call_hp_automute); spec->detect_hp = 1; } @@ -4019,9 +4036,7 @@ static int check_auto_mute_availability(struct hda_codec *codec) snd_printdd("hda-codec: Enable Line-Out auto-muting on NID 0x%x\n", nid); snd_hda_jack_detect_enable_callback(codec, nid, HDA_GEN_FRONT_EVENT, - spec->line_automute_hook ? - spec->line_automute_hook : - snd_hda_gen_line_automute); + call_line_automute); spec->detect_lo = 1; } spec->automute_lo_possible = spec->detect_hp; @@ -4063,9 +4078,7 @@ static bool auto_mic_check_imux(struct hda_codec *codec) snd_hda_jack_detect_enable_callback(codec, spec->am_entry[i].pin, HDA_GEN_MIC_EVENT, - spec->mic_autoswitch_hook ? - spec->mic_autoswitch_hook : - snd_hda_gen_mic_autoswitch); + call_mic_autoswitch); return true; } @@ -4157,7 +4170,7 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return power_state; if (get_wcaps_type(get_wcaps(codec, nid)) >= AC_WID_POWER) return power_state; - if (is_active_nid(codec, nid, HDA_OUTPUT, 0)) + if (is_active_nid_for_any(codec, nid)) return power_state; return AC_PWRST_D3; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 54e665160379..76200314ee95 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -222,6 +222,7 @@ struct hda_gen_spec { unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */ unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */ unsigned int own_eapd_ctl:1; /* set EAPD by own function */ + unsigned int keep_eapd_on:1; /* don't turn off EAPD automatically */ unsigned int vmaster_mute_enum:1; /* add vmaster mute mode enum */ unsigned int indep_hp:1; /* independent HP supported */ unsigned int prefer_hp_amp:1; /* enable HP amp for speaker if any */ diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index bd8d46cca2b3..cccaf9c7a7bb 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -58,6 +58,7 @@ enum { CS420X_GPIO_23, CS420X_MBP101, CS420X_MBP81, + CS420X_MBA42, CS420X_AUTO, /* aliases */ CS420X_IMAC27_122 = CS420X_GPIO_23, @@ -346,6 +347,7 @@ static const struct hda_model_fixup cs420x_models[] = { { .id = CS420X_APPLE, .name = "apple" }, { .id = CS420X_MBP101, .name = "mbp101" }, { .id = CS420X_MBP81, .name = "mbp81" }, + { .id = CS420X_MBA42, .name = "mba42" }, {} }; @@ -361,6 +363,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), + SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42), SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), {} /* terminator */ }; @@ -414,6 +417,20 @@ static const struct hda_pintbl mbp101_pincfgs[] = { {} /* terminator */ }; +static const struct hda_pintbl mba42_pincfgs[] = { + { 0x09, 0x012b4030 }, /* HP */ + { 0x0a, 0x400000f0 }, + { 0x0b, 0x90100120 }, /* speaker */ + { 0x0c, 0x400000f0 }, + { 0x0d, 0x90a00110 }, /* mic */ + { 0x0e, 0x400000f0 }, + { 0x0f, 0x400000f0 }, + { 0x10, 0x400000f0 }, + { 0x12, 0x400000f0 }, + { 0x15, 0x400000f0 }, + {} /* terminator */ +}; + static void cs420x_fixup_gpio_13(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -482,6 +499,12 @@ static const struct hda_fixup cs420x_fixups[] = { .chained = true, .chain_id = CS420X_GPIO_13, }, + [CS420X_MBA42] = { + .type = HDA_FIXUP_PINS, + .v.pins = mba42_pincfgs, + .chained = true, + .chain_id = CS420X_GPIO_13, + }, }; static struct cs_spec *cs_alloc_spec(struct hda_codec *codec, int vendor_nid) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6bf47f7326ad..403010c9e82e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3482,6 +3482,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05c9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ca, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05cb, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05de, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05e0, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05e9, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05ea, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05eb, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3492,6 +3494,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), @@ -3529,6 +3535,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), @@ -3592,6 +3599,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, + {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, + {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, {} }; @@ -4271,6 +4280,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE7, .name = "asus-mode7"}, {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {} }; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e0dadcf2030d..e5245544eb52 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -136,6 +136,7 @@ static struct via_spec *via_new_spec(struct hda_codec *codec) spec->codec_type = VT1708S; spec->no_pin_power_ctl = 1; spec->gen.indep_hp = 1; + spec->gen.keep_eapd_on = 1; spec->gen.pcm_playback_hook = via_playback_pcm_hook; return spec; } @@ -231,9 +232,14 @@ static void vt1708_update_hp_work(struct hda_codec *codec) static void set_widgets_power_state(struct hda_codec *codec) { +#if 0 /* FIXME: the assumed connections don't match always with the + * actual routes by the generic parser, so better to disable + * the control for safety. + */ struct via_spec *spec = codec->spec; if (spec->set_widgets_power_state) spec->set_widgets_power_state(codec); +#endif } static void update_power_state(struct hda_codec *codec, hda_nid_t nid, @@ -478,7 +484,9 @@ static int via_suspend(struct hda_codec *codec) /* Fix pop noise on headphones */ int i; for (i = 0; i < spec->gen.autocfg.hp_outs; i++) - snd_hda_set_pin_ctl(codec, spec->gen.autocfg.hp_pins[i], 0); + snd_hda_codec_write(codec, spec->gen.autocfg.hp_pins[i], + 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + 0x00); } return 0; diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index d59abe1682c5..748e82d4d257 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1341,7 +1341,8 @@ static int sis_chip_create(struct snd_card *card, if (rc) goto error_out; - if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { + rc = pci_set_dma_mask(pci, DMA_BIT_MASK(30)); + if (rc < 0) { dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA"); goto error_out_enabled; } diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h index 114f69a0c629..306d0bc8455f 100644 --- a/sound/soc/codecs/ab8500-codec.h +++ b/sound/soc/codecs/ab8500-codec.h @@ -348,25 +348,25 @@ /* AB8500_ADSLOTSELX */ #define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00 -#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x01 -#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x02 -#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x03 -#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x04 -#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x05 -#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x06 -#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x07 -#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x08 -#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0x0F +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x10 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x20 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x30 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x40 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x50 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x60 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x70 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x80 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0xF0 #define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00 -#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x10 -#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x20 -#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x30 -#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x40 -#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x50 -#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x60 -#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x70 -#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x80 -#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0xF0 +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x01 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x02 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x03 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x04 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x05 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x06 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x07 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x08 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0x0F #define AB8500_ADSLOTSELX_EVEN_SHIFT 0 #define AB8500_ADSLOTSELX_ODD_SHIFT 4 diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0f6f481cec09..987f728718c5 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -86,7 +86,7 @@ static const struct reg_default cs42l52_reg_defaults[] = { { CS42L52_BEEP_VOL, 0x00 }, /* r1D Beep Volume off Time */ { CS42L52_BEEP_TONE_CTL, 0x00 }, /* r1E Beep Tone Cfg. */ { CS42L52_TONE_CTL, 0x00 }, /* r1F Tone Ctl */ - { CS42L52_MASTERA_VOL, 0x88 }, /* r20 Master A Volume */ + { CS42L52_MASTERA_VOL, 0x00 }, /* r20 Master A Volume */ { CS42L52_MASTERB_VOL, 0x00 }, /* r21 Master B Volume */ { CS42L52_HPA_VOL, 0x00 }, /* r22 Headphone A Volume */ { CS42L52_HPB_VOL, 0x00 }, /* r23 Headphone B Volume */ @@ -193,6 +193,8 @@ static DECLARE_TLV_DB_SCALE(mic_tlv, 1600, 100, 0); static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -225,7 +227,7 @@ static const char * const mic_bias_level_text[] = { }; static const struct soc_enum mic_bias_level_enum = - SOC_ENUM_SINGLE(CS42L52_IFACE_CTL1, 0, + SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0, ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); static const char * const cs42l52_mic_text[] = { "Single", "Differential" }; @@ -260,7 +262,7 @@ static const char * const hp_gain_num_text[] = { }; static const struct soc_enum hp_gain_enum = - SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 4, + SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5, ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); static const char * const beep_pitch_text[] = { @@ -413,7 +415,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Headphone Analog Gain", hp_gain_enum), SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, - CS42L52_SPKB_VOL, 7, 0x1, 0xff, hl_tlv), + CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), @@ -441,7 +443,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, - 6, 0x7f, 0x19, hl_tlv), + 0, 0x7f, 0x19, mix_tlv), SOC_DOUBLE_R("PCM Mixer Switch", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 60985c059071..4277012c4719 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -157,7 +157,7 @@ #define CS42L52_PB_CTL1_INV_PCMA (1 << 2) #define CS42L52_PB_CTL1_MSTB_MUTE (1 << 1) #define CS42L52_PB_CTL1_MSTA_MUTE (1 << 0) -#define CS42L52_PB_CTL1_MUTE_MASK 0xFFFD +#define CS42L52_PB_CTL1_MUTE_MASK 0x03 #define CS42L52_PB_CTL1_MUTE 3 #define CS42L52_PB_CTL1_UNMUTE 0 diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 41230ad1c3e0..4a6f1daf911f 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1488,17 +1488,17 @@ static int da7213_probe(struct snd_soc_codec *codec) DA7213_DMIC_DATA_SEL_SHIFT); break; } - switch (pdata->dmic_data_sel) { + switch (pdata->dmic_samplephase) { case DA7213_DMIC_SAMPLE_ON_CLKEDGE: case DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE: - dmic_cfg |= (pdata->dmic_data_sel << + dmic_cfg |= (pdata->dmic_samplephase << DA7213_DMIC_SAMPLEPHASE_SHIFT); break; } - switch (pdata->dmic_data_sel) { + switch (pdata->dmic_clk_rate) { case DA7213_DMIC_CLK_3_0MHZ: case DA7213_DMIC_CLK_1_5MHZ: - dmic_cfg |= (pdata->dmic_data_sel << + dmic_cfg |= (pdata->dmic_clk_rate << DA7213_DMIC_CLK_RATE_SHIFT); break; } diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index ce0d36412c97..8d14a76c7249 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2233,7 +2233,7 @@ static int max98090_probe(struct snd_soc_codec *codec) dev_dbg(codec->dev, "irq = %d\n", max98090->irq); ret = request_threaded_irq(max98090->irq, NULL, - max98090_interrupt, IRQF_TRIGGER_FALLING, + max98090_interrupt, IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "max98090_interrupt", codec); if (ret < 0) { dev_err(codec->dev, "request_irq failed: %d\n", diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 65d09d60b7c6..1514bf845e4b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -187,14 +187,14 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, break; } - - if (found) - snd_soc_dapm_sync(widget->dapm); } - ret = snd_soc_update_bits(widget->codec, reg, val_mask, val); - mutex_unlock(&widget->codec->mutex); + + if (found) + snd_soc_dapm_sync(widget->dapm); + + ret = snd_soc_update_bits_locked(widget->codec, reg, val_mask, val); return ret; } diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 8df2b6e1a1a6..370af0cbcc9a 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -667,6 +667,7 @@ static int wm0010_boot(struct snd_soc_codec *codec) /* On wm0010 only the CLKCTRL1 value is used */ pll_rec.clkctrl1 = wm0010->pll_clkctrl1; + ret = -ENOMEM; len = pll_rec.length + 8; out = kzalloc(len, GFP_KERNEL); if (!out) { diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e895d3939eef..100fdadda56a 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1120,7 +1120,8 @@ SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5102_aec_loopback_mux), SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 731884e04776..88ad7db52dde 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -190,7 +190,7 @@ ARIZONA_MIXER_CONTROLS("DSP2R", ARIZONA_DSP2RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3L", ARIZONA_DSP3LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP3R", ARIZONA_DSP3RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DSP4L", ARIZONA_DSP4LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DSP5R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DSP4R", ARIZONA_DSP4RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), @@ -503,7 +503,8 @@ SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, - ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5110_aec_loopback_mux), + ARIZONA_AEC_LOOPBACK_ENA_SHIFT, 0, + &wm5110_aec_loopback_mux), SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), @@ -976,6 +977,8 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; + arizona_init_spk(codec); + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1eb152cb1097..29e95f93d482 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -383,6 +383,8 @@ static int wm8994_get_drc_enum(struct snd_kcontrol *kcontrol, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int drc = wm8994_get_drc(kcontrol->id.name); + if (drc < 0) + return drc; ucontrol->value.enumerated.item[0] = wm8994->drc_cfg[drc]; return 0; @@ -488,6 +490,9 @@ static int wm8994_get_retune_mobile_enum(struct snd_kcontrol *kcontrol, struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); int block = wm8994_get_retune_mobile_block(kcontrol->id.name); + if (block < 0) + return block; + ucontrol->value.enumerated.item[0] = wm8994->retune_mobile_cfg[block]; return 0; @@ -1031,7 +1036,7 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = codec->control_data; + struct wm8994 *control = wm8994->wm8994; int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; int i; int dac; @@ -3831,8 +3836,14 @@ static irqreturn_t wm8958_mic_irq(int irq, void *data) ret); } else if (!(ret & WM1811_JACKDET_LVL)) { dev_dbg(codec->dev, "Ignoring removed jack\n"); - return IRQ_HANDLED; + goto out; } + } else if (!(reg & WM8958_MICD_STS)) { + snd_soc_jack_report(wm8994->micdet[0].jack, 0, + SND_JACK_MECHANICAL | SND_JACK_HEADSET | + wm8994->btn_mask); + wm8994->mic_detecting = true; + goto out; } if (wm8994->mic_detecting) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 56ecfc72f2e9..81490febac6d 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -631,7 +631,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (word_length / 4) & 0x7; + u32 tx_rotate = (word_length / 4) & 0x7; + u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; /* @@ -655,9 +656,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(rotate), TXROT(7)); + TXROT(tx_rotate), TXROT(7)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXROT(rotate), RXROT(7)); + RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); } diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 902fab02b851..c6fa03e2114a 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -540,11 +540,6 @@ static int imx_ssi_probe(struct platform_device *pdev) clk_prepare_enable(ssi->clk); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - ret = -ENODEV; - goto failed_get_resource; - } - ssi->base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(ssi->base)) { ret = PTR_ERR(ssi->base); @@ -633,7 +628,6 @@ failed_pdev_fiq_alloc: snd_soc_unregister_component(&pdev->dev); failed_register: release_mem_region(res->start, resource_size(res)); -failed_get_resource: clk_disable_unprepare(ssi->clk); failed_clk: diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index befe68f59285..4c9dad3263c5 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -471,11 +471,6 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, priv); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "platform_get_resource failed\n"); - return -ENXIO; - } - priv->io = devm_ioremap_resource(&pdev->dev, mem); if (IS_ERR(priv->io)) return PTR_ERR(priv->io); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7eb3f28..06a8000aa07b 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a80c883bb8be..c7051c457b75 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -55,7 +55,8 @@ static int dapm_up_seq[] = { [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, - [snd_soc_dapm_dai] = 3, + [snd_soc_dapm_dai_in] = 3, + [snd_soc_dapm_dai_out] = 3, [snd_soc_dapm_aif_in] = 3, [snd_soc_dapm_aif_out] = 3, [snd_soc_dapm_mic] = 4, @@ -92,7 +93,8 @@ static int dapm_down_seq[] = { [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, - [snd_soc_dapm_dai] = 10, + [snd_soc_dapm_dai_in] = 10, + [snd_soc_dapm_dai_out] = 10, [snd_soc_dapm_dai_link] = 11, [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, @@ -419,7 +421,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_hp: case snd_soc_dapm_mic: case snd_soc_dapm_spk: @@ -820,7 +823,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_out: if (widget->active) { widget->outputs = snd_soc_dapm_suspend_check(widget); return widget->outputs; @@ -916,7 +919,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: if (widget->active) { widget->inputs = snd_soc_dapm_suspend_check(widget); return widget->inputs; @@ -1135,16 +1138,6 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) return out != 0 && in != 0; } -static int dapm_dai_check_power(struct snd_soc_dapm_widget *w) -{ - DAPM_UPDATE_STAT(w, power_checks); - - if (w->active) - return w->active; - - return dapm_generic_check_power(w); -} - /* Check to see if an ADC has power */ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { @@ -2318,7 +2311,8 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai: + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: case snd_soc_dapm_dai_link: list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &wsink->sources); @@ -3129,10 +3123,12 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: + case snd_soc_dapm_dai_out: w->power_check = dapm_adc_check_power; break; case snd_soc_dapm_dac: case snd_soc_dapm_aif_in: + case snd_soc_dapm_dai_in: w->power_check = dapm_dac_check_power; break; case snd_soc_dapm_pga: @@ -3152,9 +3148,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; - case snd_soc_dapm_dai: - w->power_check = dapm_dai_check_power; - break; default: w->power_check = dapm_always_on_check_power; break; @@ -3375,7 +3368,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.reg = SND_SOC_NOPM; if (dai->driver->playback.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_in; template.name = dai->driver->playback.stream_name; template.sname = dai->driver->playback.stream_name; @@ -3393,7 +3386,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, } if (dai->driver->capture.stream_name) { - template.id = snd_soc_dapm_dai; + template.id = snd_soc_dapm_dai_out; template.name = dai->driver->capture.stream_name; template.sname = dai->driver->capture.stream_name; @@ -3423,8 +3416,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) /* For each DAI widget... */ list_for_each_entry(dai_w, &card->widgets, list) { - if (dai_w->id != snd_soc_dapm_dai) + switch (dai_w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } dai = dai_w->priv; @@ -3433,8 +3431,13 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) if (w->dapm != dai_w->dapm) continue; - if (w->id == snd_soc_dapm_dai) + switch (w->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: continue; + default: + break; + } if (!w->sname) continue; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 73bb8eefa491..ccb6be4d658d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -928,8 +928,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, /* Create any new FE <--> BE connections */ for (i = 0; i < list->num_widgets; i++) { - if (list->widgets[i]->id != snd_soc_dapm_dai) + switch (list->widgets[i]->id) { + case snd_soc_dapm_dai_in: + case snd_soc_dapm_dai_out: + break; + default: continue; + } /* is there a valid BE rtd for this widget */ be = dpcm_get_be(card, list->widgets[i], stream); @@ -2011,9 +2016,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) capture = 1; } diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index a1d9b0792a1e..b9defcdeb7ef 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -42,8 +42,8 @@ static const u8 ep_w_max_packet_size[] = { 0x94, 0x01, 0x5c, 0x02 /* alt 3: 404 EP2 and 604 EP6 (25 fpp) */ }; -static const u8 known_fw_versions[][4] = { - { 0x03, 0x01, 0x0b, 0x00 } +static const u8 known_fw_versions[][2] = { + { 0x03, 0x01 } }; struct ihex_record { @@ -343,7 +343,7 @@ static int usb6fire_fw_check(u8 *version) int i; for (i = 0; i < ARRAY_SIZE(known_fw_versions); i++) - if (!memcmp(version, known_fw_versions + i, 4)) + if (!memcmp(version, known_fw_versions + i, 2)) return 0; snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. " diff --git a/sound/usb/card.c b/sound/usb/card.c index 1a033177b83f..64952e2d3ed1 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -147,14 +147,32 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int return -EINVAL; } + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + + /* + * Android with both accessory and audio interfaces enabled gets the + * interface numbers wrong. + */ + if ((chip->usb_id == USB_ID(0x18d1, 0x2d04) || + chip->usb_id == USB_ID(0x18d1, 0x2d05)) && + interface == 0 && + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && + altsd->bInterfaceSubClass == USB_SUBCLASS_VENDOR_SPEC) { + interface = 2; + iface = usb_ifnum_to_if(dev, interface); + if (!iface) + return -EINVAL; + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + } + if (usb_interface_claimed(iface)) { snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", dev->devnum, ctrlif, interface); return -EINVAL; } - alts = &iface->altsetting[0]; - altsd = get_iface_desc(alts); if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) { diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index ca4739c3f650..d5438083fd6a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -885,7 +885,9 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x0808): case USB_ID(0x046d, 0x0809): + case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ + case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. diff --git a/sound/usb/proc.c b/sound/usb/proc.c index 135c76871063..5f761ab34c01 100644 --- a/sound/usb/proc.c +++ b/sound/usb/proc.c @@ -116,21 +116,22 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } static void proc_dump_ep_status(struct snd_usb_substream *subs, - struct snd_usb_endpoint *ep, + struct snd_usb_endpoint *data_ep, + struct snd_usb_endpoint *sync_ep, struct snd_info_buffer *buffer) { - if (!ep) + if (!data_ep) return; - snd_iprintf(buffer, " Packet Size = %d\n", ep->curpacksize); + snd_iprintf(buffer, " Packet Size = %d\n", data_ep->curpacksize); snd_iprintf(buffer, " Momentary freq = %u Hz (%#x.%04x)\n", subs->speed == USB_SPEED_FULL - ? get_full_speed_hz(ep->freqm) - : get_high_speed_hz(ep->freqm), - ep->freqm >> 16, ep->freqm & 0xffff); - if (ep->freqshift != INT_MIN) { - int res = 16 - ep->freqshift; + ? get_full_speed_hz(data_ep->freqm) + : get_high_speed_hz(data_ep->freqm), + data_ep->freqm >> 16, data_ep->freqm & 0xffff); + if (sync_ep && data_ep->freqshift != INT_MIN) { + int res = 16 - data_ep->freqshift; snd_iprintf(buffer, " Feedback Format = %d.%d\n", - (ep->syncmaxsize > 3 ? 32 : 24) - res, res); + (sync_ep->syncmaxsize > 3 ? 32 : 24) - res, res); } } @@ -140,8 +141,7 @@ static void proc_dump_substream_status(struct snd_usb_substream *subs, struct sn snd_iprintf(buffer, " Status: Running\n"); snd_iprintf(buffer, " Interface = %d\n", subs->interface); snd_iprintf(buffer, " Altset = %d\n", subs->altset_idx); - proc_dump_ep_status(subs, subs->data_endpoint, buffer); - proc_dump_ep_status(subs, subs->sync_endpoint, buffer); + proc_dump_ep_status(subs, subs->data_endpoint, subs->sync_endpoint, buffer); } else { snd_iprintf(buffer, " Status: Stop\n"); } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 7f1722f82c89..8b75bcf136f6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -215,7 +215,13 @@ .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { - USB_DEVICE(0x046d, 0x0990), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, + .idProduct = 0x0990, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Logitech, Inc.", .product_name = "QuickCam Pro 9000", @@ -1792,7 +1798,11 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_VENDOR_SPEC(0x0582, 0x0108), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = 0, - .type = QUIRK_MIDI_STANDARD_INTERFACE + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0007, + .in_cables = 0x0007 + } } }, { |