From 1a39b5e1f932b0ab292c1737724f17bd6a73d630 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 14:32:16 +0200 Subject: ALSA: hda - Add GPIO control to AD1884 HP fixup The AD1884 HP laptop/mobile quirks control GPIO1 bit as the primary mute as well. Add the similar control to ad1884 fixup for auto parser, too. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d97f0d61a15b..2ae7dc54ab1d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3599,14 +3599,34 @@ static void ad1884_fixup_amp_override(struct hda_codec *codec, (1 << AC_AMPCAP_MUTE_SHIFT)); } +/* toggle GPIO1 according to the mute state */ +static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + struct ad198x_spec *spec = codec->spec; + + if (spec->eapd_nid) + ad_vmaster_eapd_hook(private_data, enabled); + snd_hda_codec_update_cache(codec, 0x01, 0, + AC_VERB_SET_GPIO_DATA, + enabled ? 0x00 : 0x02); +} + static void ad1884_fixup_hp_eapd(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct ad198x_spec *spec = codec->spec; + static const struct hda_verb gpio_init_verbs[] = { + {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, + {}, + }; switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; + spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook; + snd_hda_sequence_write_cache(codec, gpio_init_verbs); break; case HDA_FIXUP_ACT_PROBE: if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) -- cgit v1.2.3 From 6a699bec88d5755c0f1be4e967649b3cfeac0205 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 14:45:37 +0200 Subject: ALSA: hda - Add fixup for Lenovo Thinkpad with AD1984 codec Ported from the static quirk (model=thinkpad). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2ae7dc54ab1d..0262ffb96538 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3637,9 +3637,17 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, } } +/* set magic COEFs for dmic */ +static const struct hda_verb ad1884_dmic_init_verbs[] = { + {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, + {0x01, AC_VERB_SET_PROC_COEF, 0x08}, + {} +}; + enum { AD1884_FIXUP_AMP_OVERRIDE, AD1884_FIXUP_HP_EAPD, + AD1884_FIXUP_DMIC_COEF, }; static const struct hda_fixup ad1884_fixups[] = { @@ -3653,10 +3661,15 @@ static const struct hda_fixup ad1884_fixups[] = { .chained = true, .chain_id = AD1884_FIXUP_AMP_OVERRIDE, }, + [AD1884_FIXUP_DMIC_COEF] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = ad1884_dmic_init_verbs, + }, }; static const struct snd_pci_quirk ad1884_fixup_tbl[] = { SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF), {} }; -- cgit v1.2.3 From f404627d27b27d79287dee7c6dba934790959ee3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 15:14:17 +0200 Subject: ALSA: hda - Add fixup for HP TouchSmart with AD1984A codec Ported from the static quirk. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0262ffb96538..a667256984fd 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3648,6 +3648,7 @@ enum { AD1884_FIXUP_AMP_OVERRIDE, AD1884_FIXUP_HP_EAPD, AD1884_FIXUP_DMIC_COEF, + AD1884_FIXUP_HP_TOUCHSMART, }; static const struct hda_fixup ad1884_fixups[] = { @@ -3665,9 +3666,16 @@ static const struct hda_fixup ad1884_fixups[] = { .type = HDA_FIXUP_VERBS, .v.verbs = ad1884_dmic_init_verbs, }, + [AD1884_FIXUP_HP_TOUCHSMART] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = ad1884_dmic_init_verbs, + .chained = true, + .chain_id = AD1884_FIXUP_HP_EAPD, + }, }; static const struct snd_pci_quirk ad1884_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART), SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF), {} -- cgit v1.2.3 From aa95d61b43e0fcb0b2ce68e5efa37174fd9e5cd3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 15:16:31 +0200 Subject: ALSA: hda - Remove static quirks for AD1882 Now the generic parser can work stably enough, we can get rid of the static quirks. Let's start from AD1882. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 384 +------------------------------------------ 1 file changed, 1 insertion(+), 383 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index a667256984fd..876d836ef742 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -4891,299 +4891,7 @@ static int patch_ad1884a(struct hda_codec *codec) * port-G - rear clfe-out (6stack) */ -#ifdef ENABLE_AD_STATIC_QUIRKS -static const hda_nid_t ad1882_dac_nids[3] = { - 0x04, 0x03, 0x05 -}; - -static const hda_nid_t ad1882_adc_nids[2] = { - 0x08, 0x09, -}; - -static const hda_nid_t ad1882_capsrc_nids[2] = { - 0x0c, 0x0d, -}; - -#define AD1882_SPDIF_OUT 0x02 - -/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */ -static const struct hda_input_mux ad1882_capture_source = { - .num_items = 5, - .items = { - { "Front Mic", 0x1 }, - { "Mic", 0x4 }, - { "Line", 0x2 }, - { "CD", 0x3 }, - { "Mix", 0x7 }, - }, -}; - -/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */ -static const struct hda_input_mux ad1882a_capture_source = { - .num_items = 5, - .items = { - { "Front Mic", 0x1 }, - { "Mic", 0x4}, - { "Line", 0x2 }, - { "Digital Mic", 0x06 }, - { "Mix", 0x7 }, - }, -}; - -static const struct snd_kcontrol_new ad1882_base_mixers[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line-In Boost Volume", 0x3a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* SPDIF controls */ - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - /* identical with ad1983 */ - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1882_loopback_mixers[] = { - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1882a_loopback_mixers[] = { - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Digital Mic Boost Volume", 0x1f, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1882_3stack_mixers[] = { - HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = ad198x_ch_mode_info, - .get = ad198x_ch_mode_get, - .put = ad198x_ch_mode_put, - }, - { } /* end */ -}; - -/* simple auto-mute control for AD1882 3-stack board */ -#define AD1882_HP_EVENT 0x01 - -static void ad1882_3stack_automute(struct hda_codec *codec) -{ - bool mute = snd_hda_jack_detect(codec, 0x11); - snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, - mute ? 0 : PIN_OUT); -} - -static int ad1882_3stack_automute_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1882_3stack_automute(codec); - return 0; -} - -static void ad1882_3stack_unsol_event(struct hda_codec *codec, unsigned int res) -{ - switch (res >> 26) { - case AD1882_HP_EVENT: - ad1882_3stack_automute(codec); - break; - } -} - -static const struct snd_kcontrol_new ad1882_6stack_mixers[] = { - HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct hda_verb ad1882_ch2_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - { } /* end */ -}; - -static const struct hda_verb ad1882_ch4_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - { } /* end */ -}; - -static const struct hda_verb ad1882_ch6_init[] = { - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - { } /* end */ -}; - -static const struct hda_channel_mode ad1882_modes[3] = { - { 2, ad1882_ch2_init }, - { 4, ad1882_ch4_init }, - { 6, ad1882_ch6_init }, -}; - -/* - * initialization verbs - */ -static const struct hda_verb ad1882_init_verbs[] = { - /* DACs; mute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-A (HP) mixer */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* HP selector - select DAC2 */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Port-D (Line-out) mixer */ - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-D pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mono-out mixer */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Mono-out pin */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Port-B (front mic) pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ - /* Port-C (line-in) pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ - /* Port-C mixer - mute as input */ - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Port-E (mic-in) pin */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ - /* Port-E mixer - mute as input */ - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Port-F (surround) */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Port-G (CLFE) */ - {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Analog mixer; mute as default */ - /* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - /* SPDIF output selector */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - { } /* end */ -}; - -static const struct hda_verb ad1882_3stack_automute_verbs[] = { - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1882_HP_EVENT}, - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1882_loopbacks[] = { - { 0x20, HDA_INPUT, 0 }, /* Front Mic */ - { 0x20, HDA_INPUT, 1 }, /* Mic */ - { 0x20, HDA_INPUT, 4 }, /* Line */ - { 0x20, HDA_INPUT, 6 }, /* CD */ - { } /* end */ -}; -#endif - -/* models */ -enum { - AD1882_AUTO, - AD1882_3STACK, - AD1882_6STACK, - AD1882_3STACK_AUTOMUTE, - AD1882_MODELS -}; - -static const char * const ad1882_models[AD1986A_MODELS] = { - [AD1882_AUTO] = "auto", - [AD1882_3STACK] = "3stack", - [AD1882_6STACK] = "6stack", - [AD1882_3STACK_AUTOMUTE] = "3stack-automute", -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ - -static int ad1882_parse_auto_config(struct hda_codec *codec) +static int patch_ad1882(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -5210,96 +4918,6 @@ static int ad1882_parse_auto_config(struct hda_codec *codec) return err; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1882(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1882_MODELS, - ad1882_models, NULL); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1882_AUTO; - } - - if (board_config == AD1882_AUTO) - return ad1882_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 6; - spec->multiout.num_dacs = 3; - spec->multiout.dac_nids = ad1882_dac_nids; - spec->multiout.dig_out_nid = AD1882_SPDIF_OUT; - spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); - spec->adc_nids = ad1882_adc_nids; - spec->capsrc_nids = ad1882_capsrc_nids; - if (codec->vendor_id == 0x11d41882) - spec->input_mux = &ad1882_capture_source; - else - spec->input_mux = &ad1882a_capture_source; - spec->num_mixers = 2; - spec->mixers[0] = ad1882_base_mixers; - if (codec->vendor_id == 0x11d41882) - spec->mixers[1] = ad1882_loopback_mixers; - else - spec->mixers[1] = ad1882a_loopback_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1882_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1882_loopbacks; -#endif - spec->vmaster_nid = 0x04; - - codec->patch_ops = ad198x_patch_ops; - - /* override some parameters */ - switch (board_config) { - default: - case AD1882_3STACK: - case AD1882_3STACK_AUTOMUTE: - spec->num_mixers = 3; - spec->mixers[2] = ad1882_3stack_mixers; - spec->channel_mode = ad1882_modes; - spec->num_channel_mode = ARRAY_SIZE(ad1882_modes); - spec->need_dac_fix = 1; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - if (board_config != AD1882_3STACK) { - spec->init_verbs[spec->num_init_verbs++] = - ad1882_3stack_automute_verbs; - codec->patch_ops.unsol_event = ad1882_3stack_unsol_event; - codec->patch_ops.init = ad1882_3stack_automute_init; - } - break; - case AD1882_6STACK: - spec->num_mixers = 3; - spec->mixers[2] = ad1882_6stack_mixers; - break; - } - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1882 ad1882_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * patch entries -- cgit v1.2.3 From 5ccc618fee67f0f0b2122dd4b32a02fd2b6a1569 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 15:36:56 +0200 Subject: ALSA: hda - Remove static quirks for AD1884/1984 & variants Since the necessary device-specific fixups for Thinkpad and HP devices have been already ported, we can remove all static quirks for AD1884, AD1984, AD1884A and AD1984A codecs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1348 +----------------------------------------- 1 file changed, 18 insertions(+), 1330 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 876d836ef742..bfa8f532841d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3423,167 +3423,19 @@ static int patch_ad1988(struct hda_codec *codec) * * AD1984 = AD1884 + two digital mic-ins * - * FIXME: - * For simplicity, we share the single DAC for both HP and line-outs - * right now. The inidividual playbacks could be easily implemented, - * but no build-up framework is given, so far. - */ - -#ifdef ENABLE_AD_STATIC_QUIRKS -static const hda_nid_t ad1884_dac_nids[1] = { - 0x04, -}; - -static const hda_nid_t ad1884_adc_nids[2] = { - 0x08, 0x09, -}; - -static const hda_nid_t ad1884_capsrc_nids[2] = { - 0x0c, 0x0d, -}; - -#define AD1884_SPDIF_OUT 0x02 - -static const struct hda_input_mux ad1884_capture_source = { - .num_items = 4, - .items = { - { "Front Mic", 0x0 }, - { "Mic", 0x1 }, - { "CD", 0x2 }, - { "Mix", 0x3 }, - }, -}; - -static const struct snd_kcontrol_new ad1884_base_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* SPDIF controls */ - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - /* identical with ad1983 */ - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1984_dmic_mixers[] = { - HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0, - HDA_INPUT), - HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0, - HDA_INPUT), - { } /* end */ -}; - -/* - * initialization verbs + * AD1883 / AD1884A / AD1984A / AD1984B + * + * port-B (0x14) - front mic-in + * port-E (0x1c) - rear mic-in + * port-F (0x16) - CD / ext out + * port-C (0x15) - rear line-in + * port-D (0x12) - rear line-out + * port-A (0x11) - front hp-out + * + * AD1984A = AD1884A + digital-mic + * AD1883 = equivalent with AD1984A + * AD1984B = AD1984A + extra SPDIF-out */ -static const struct hda_verb ad1884_init_verbs[] = { - /* DACs; mute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-A (HP) mixer */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* HP selector - select DAC2 */ - {0x22, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Port-D (Line-out) mixer */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-D pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mono-out mixer */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Mono-out pin */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mono selector */ - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Port-B (front mic) pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Port-C (rear mic) pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - /* SPDIF output selector */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1884_loopbacks[] = { - { 0x20, HDA_INPUT, 0 }, /* Front Mic */ - { 0x20, HDA_INPUT, 1 }, /* Mic */ - { 0x20, HDA_INPUT, 2 }, /* CD */ - { 0x20, HDA_INPUT, 4 }, /* Docking */ - { } /* end */ -}; -#endif - -static const char * const ad1884_slave_vols[] = { - "PCM", "Mic", "Mono", "Front Mic", "Mic", "CD", - "Internal Mic", "Dock Mic", /* "Beep", */ "IEC958", - NULL -}; - -enum { - AD1884_AUTO, - AD1884_BASIC, - AD1884_MODELS -}; - -static const char * const ad1884_models[AD1884_MODELS] = { - [AD1884_AUTO] = "auto", - [AD1884_BASIC] = "basic", -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* set the upper-limit for mixer amp to 0dB for avoiding the possible * damage by overloading @@ -3682,7 +3534,7 @@ static const struct snd_pci_quirk ad1884_fixup_tbl[] = { }; -static int ad1884_parse_auto_config(struct hda_codec *codec) +static int patch_ad1884(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -3715,1170 +3567,6 @@ static int ad1884_parse_auto_config(struct hda_codec *codec) return err; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1884_basic(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err; - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids); - spec->multiout.dac_nids = ad1884_dac_nids; - spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; - spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids); - spec->adc_nids = ad1884_adc_nids; - spec->capsrc_nids = ad1884_capsrc_nids; - spec->input_mux = &ad1884_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1884_base_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1884_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1884_loopbacks; -#endif - spec->vmaster_nid = 0x04; - /* we need to cover all playback volumes */ - spec->slave_vols = ad1884_slave_vols; - /* slaves may contain input volumes, so we can't raise to 0dB blindly */ - spec->avoid_init_slave_vol = 1; - - codec->patch_ops = ad198x_patch_ops; - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} - -static int patch_ad1884(struct hda_codec *codec) -{ - int board_config; - - board_config = snd_hda_check_board_config(codec, AD1884_MODELS, - ad1884_models, NULL); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1884_AUTO; - } - - if (board_config == AD1884_AUTO) - return ad1884_parse_auto_config(codec); - else - return patch_ad1884_basic(codec); -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1884 ad1884_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - -#ifdef ENABLE_AD_STATIC_QUIRKS -/* - * Lenovo Thinkpad T61/X61 - */ -static const struct hda_input_mux ad1984_thinkpad_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x1 }, - { "Mix", 0x3 }, - { "Dock Mic", 0x4 }, - }, -}; - - -/* - * Dell Precision T3400 - */ -static const struct hda_input_mux ad1984_dell_desktop_capture_source = { - .num_items = 3, - .items = { - { "Front Mic", 0x0 }, - { "Line-In", 0x1 }, - { "Mix", 0x3 }, - }, -}; - - -static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* SPDIF controls */ - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - /* identical with ad1983 */ - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -/* additional verbs */ -static const struct hda_verb ad1984_thinkpad_init_verbs[] = { - /* Port-E (docking station mic) pin */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* docking mic boost */ - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Analog PC Beeper - allow firmware/ACPI beeps */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3) | 0x1a}, - /* Analog mixer - docking mic; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* enable EAPD bit */ - {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - { } /* end */ -}; - -/* - * Dell Precision T3400 - */ -static const struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Line-In Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -/* Digial MIC ADC NID 0x05 + 0x06 */ -static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - snd_hda_codec_setup_stream(codec, 0x05 + substream->number, - stream_tag, 0, format); - return 0; -} - -static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number); - return 0; -} - -static const struct hda_pcm_stream ad1984_pcm_dmic_capture = { - .substreams = 2, - .channels_min = 2, - .channels_max = 2, - .nid = 0x05, - .ops = { - .prepare = ad1984_pcm_dmic_prepare, - .cleanup = ad1984_pcm_dmic_cleanup - }, -}; - -static int ad1984_build_pcms(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - struct hda_pcm *info; - int err; - - err = ad198x_build_pcms(codec); - if (err < 0) - return err; - - info = spec->pcm_rec + codec->num_pcms; - codec->num_pcms++; - info->name = "AD1984 Digital Mic"; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture; - return 0; -} - -/* models */ -enum { - AD1984_AUTO, - AD1984_BASIC, - AD1984_THINKPAD, - AD1984_DELL_DESKTOP, - AD1984_MODELS -}; - -static const char * const ad1984_models[AD1984_MODELS] = { - [AD1984_AUTO] = "auto", - [AD1984_BASIC] = "basic", - [AD1984_THINKPAD] = "thinkpad", - [AD1984_DELL_DESKTOP] = "dell_desktop", -}; - -static const struct snd_pci_quirk ad1984_cfg_tbl[] = { - /* Lenovo Thinkpad T61/X61 */ - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), - SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), - SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP), - {} -}; - -static int patch_ad1984(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int board_config, err; - - board_config = snd_hda_check_board_config(codec, AD1984_MODELS, - ad1984_models, ad1984_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1984_AUTO; - } - - if (board_config == AD1984_AUTO) - return ad1884_parse_auto_config(codec); - - err = patch_ad1884_basic(codec); - if (err < 0) - return err; - spec = codec->spec; - - switch (board_config) { - case AD1984_BASIC: - /* additional digital mics */ - spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers; - codec->patch_ops.build_pcms = ad1984_build_pcms; - break; - case AD1984_THINKPAD: - if (codec->subsystem_id == 0x17aa20fb) { - /* Thinpad X300 does not have the ability to do SPDIF, - or attach to docking station to use SPDIF */ - spec->multiout.dig_out_nid = 0; - } else - spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; - spec->input_mux = &ad1984_thinkpad_capture_source; - spec->mixers[0] = ad1984_thinkpad_mixers; - spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs; - spec->analog_beep = 1; - break; - case AD1984_DELL_DESKTOP: - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1984_dell_desktop_capture_source; - spec->mixers[0] = ad1984_dell_desktop_mixers; - break; - } - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1984 ad1884_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - -/* - * AD1883 / AD1884A / AD1984A / AD1984B - * - * port-B (0x14) - front mic-in - * port-E (0x1c) - rear mic-in - * port-F (0x16) - CD / ext out - * port-C (0x15) - rear line-in - * port-D (0x12) - rear line-out - * port-A (0x11) - front hp-out - * - * AD1984A = AD1884A + digital-mic - * AD1883 = equivalent with AD1984A - * AD1984B = AD1984A + extra SPDIF-out - * - * FIXME: - * We share the single DAC for both HP and line-outs (see AD1884/1984). - */ - -#ifdef ENABLE_AD_STATIC_QUIRKS -static const hda_nid_t ad1884a_dac_nids[1] = { - 0x03, -}; - -#define ad1884a_adc_nids ad1884_adc_nids -#define ad1884a_capsrc_nids ad1884_capsrc_nids - -#define AD1884A_SPDIF_OUT 0x02 - -static const struct hda_input_mux ad1884a_capture_source = { - .num_items = 5, - .items = { - { "Front Mic", 0x0 }, - { "Mic", 0x4 }, - { "Line", 0x1 }, - { "CD", 0x2 }, - { "Mix", 0x3 }, - }, -}; - -static const struct snd_kcontrol_new ad1884a_base_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 2, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* SPDIF controls */ - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - /* identical with ad1983 */ - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -/* - * initialization verbs - */ -static const struct hda_verb ad1884a_init_verbs[] = { - /* DACs; unmute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - /* Port-A (HP) mixer - route only from analog mixer */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Port-D (Line-out) mixer - route only from analog mixer */ - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-D pin */ - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Mono-out mixer - route only from analog mixer */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Mono-out pin */ - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Port-B (front mic) pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Port-C (rear line-in) pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Port-E (rear mic) pin */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */ - /* Port-F (CD) pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* capture sources */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* SPDIF output amp */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1884a_loopbacks[] = { - { 0x20, HDA_INPUT, 0 }, /* Front Mic */ - { 0x20, HDA_INPUT, 1 }, /* Mic */ - { 0x20, HDA_INPUT, 2 }, /* CD */ - { 0x20, HDA_INPUT, 4 }, /* Docking */ - { } /* end */ -}; -#endif - -/* - * Laptop model - * - * Port A: Headphone jack - * Port B: MIC jack - * Port C: Internal MIC - * Port D: Dock Line Out (if enabled) - * Port E: Dock Line In (if enabled) - * Port F: Internal speakers - */ - -static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); - int mute = (!ucontrol->value.integer.value[0] && - !ucontrol->value.integer.value[1]); - /* toggle GPIO1 according to the mute state */ - snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, - mute ? 0x02 : 0x0); - return ret; -} - -static const struct snd_kcontrol_new ad1884a_laptop_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = ad1884a_mobile_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), - }, - HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1884a_mobile_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = ad1884a_mobile_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), - }, - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -/* mute internal speaker if HP is plugged */ -static void ad1884a_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x11); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, - present ? 0x00 : 0x02); -} - -/* switch to external mic if plugged */ -static void ad1884a_hp_automic(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x14); - snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, - present ? 0 : 1); -} - -#define AD1884A_HP_EVENT 0x37 -#define AD1884A_MIC_EVENT 0x36 - -/* unsolicited event for HP jack sensing */ -static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res) -{ - switch (res >> 26) { - case AD1884A_HP_EVENT: - ad1884a_hp_automute(codec); - break; - case AD1884A_MIC_EVENT: - ad1884a_hp_automic(codec); - break; - } -} - -/* initialize jack-sensing, too */ -static int ad1884a_hp_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1884a_hp_automute(codec); - ad1884a_hp_automic(codec); - return 0; -} - -/* mute internal speaker if HP or docking HP is plugged */ -static void ad1884a_laptop_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x11); - if (!present) - present = snd_hda_jack_detect(codec, 0x12); - snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); - snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE, - present ? 0x00 : 0x02); -} - -/* switch to external mic if plugged */ -static void ad1884a_laptop_automic(struct hda_codec *codec) -{ - unsigned int idx; - - if (snd_hda_jack_detect(codec, 0x14)) - idx = 0; - else if (snd_hda_jack_detect(codec, 0x1c)) - idx = 4; - else - idx = 1; - snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx); -} - -/* unsolicited event for HP jack sensing */ -static void ad1884a_laptop_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case AD1884A_HP_EVENT: - ad1884a_laptop_automute(codec); - break; - case AD1884A_MIC_EVENT: - ad1884a_laptop_automic(codec); - break; - } -} - -/* initialize jack-sensing, too */ -static int ad1884a_laptop_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1884a_laptop_automute(codec); - ad1884a_laptop_automic(codec); - return 0; -} - -/* additional verbs for laptop model */ -static const struct hda_verb ad1884a_laptop_verbs[] = { - /* Port-A (HP) pin - always unmuted */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-F (int speaker) mixer - route only from analog mixer */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-F (int speaker) pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* required for compaq 6530s/6531s speaker output */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-C pin - internal mic-in */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - /* Port-D (docking line-out) pin - default unmuted */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, - /* allow to touch GPIO1 (for mute control) */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ - { } /* end */ -}; - -static const struct hda_verb ad1884a_mobile_verbs[] = { - /* DACs; unmute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - /* Port-A (HP) mixer - route only from analog mixer */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Port-A (HP) pin - always unmuted */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-B (mic jack) pin */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - /* Port-C (int mic) pin */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - /* Port-F (int speaker) mixer - route only from analog mixer */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-F pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* capture sources */ - /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, - /* allow to touch GPIO1 (for mute control) */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ - { } /* end */ -}; - -/* - * Thinkpad X300 - * 0x11 - HP - * 0x12 - speaker - * 0x14 - mic-in - * 0x17 - built-in mic - */ - -static const struct hda_verb ad1984a_thinkpad_verbs[] = { - /* HP unmute */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* turn on EAPD */ - {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - /* internal mic - dmic */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* set magic COEFs for dmic */ - {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, - {0x01, AC_VERB_SET_PROC_COEF, 0x08}, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -static const struct hda_input_mux ad1984a_thinkpad_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x5 }, - { "Mix", 0x3 }, - }, -}; - -/* mute internal speaker if HP is plugged */ -static void ad1984a_thinkpad_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x11); - snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - -/* unsolicited event for HP jack sensing */ -static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != AD1884A_HP_EVENT) - return; - ad1984a_thinkpad_automute(codec); -} - -/* initialize jack-sensing, too */ -static int ad1984a_thinkpad_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1984a_thinkpad_automute(codec); - return 0; -} - -/* - * Precision R5500 - * 0x12 - HP/line-out - * 0x13 - speaker (mono) - * 0x15 - mic-in - */ - -static const struct hda_verb ad1984a_precision_verbs[] = { - /* Unmute main output path */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */ - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - /* Select mic as input */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */ - /* Configure as mic */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ - /* HP unmute */ - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* turn on EAPD */ - {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, - /* unsolicited event for pin-sense */ - {0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1984a_precision_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - { } /* end */ -}; - - -/* mute internal speaker if HP is plugged */ -static void ad1984a_precision_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x12); - snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - - -/* unsolicited event for HP jack sensing */ -static void ad1984a_precision_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != AD1884A_HP_EVENT) - return; - ad1984a_precision_automute(codec); -} - -/* initialize jack-sensing, too */ -static int ad1984a_precision_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1984a_precision_automute(codec); - return 0; -} - - -/* - * HP Touchsmart - * port-A (0x11) - front hp-out - * port-B (0x14) - unused - * port-C (0x15) - unused - * port-D (0x12) - rear line out - * port-E (0x1c) - front mic-in - * port-F (0x16) - Internal speakers - * digital-mic (0x17) - Internal mic - */ - -static const struct hda_verb ad1984a_touchsmart_verbs[] = { - /* DACs; unmute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */ - /* Port-A (HP) mixer - route only from analog mixer */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-A pin */ - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Port-A (HP) pin - always unmuted */ - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-E (int speaker) mixer - route only from analog mixer */ - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03}, - /* Port-E pin */ - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - /* Port-F (int speaker) mixer - route only from analog mixer */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - /* Port-F pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* Analog mixer; mute as default */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* capture sources */ - /* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT}, - {0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT}, - /* allow to touch GPIO1 (for mute control) */ - {0x01, AC_VERB_SET_GPIO_MASK, 0x02}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */ - /* internal mic - dmic */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* set magic COEFs for dmic */ - {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, - {0x01, AC_VERB_SET_PROC_COEF, 0x08}, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), -/* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_AMP_FLAG, - .name = "Master Playback Switch", - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = ad1884a_mobile_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), - }, - HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT), - { } /* end */ -}; - -/* switch to external mic if plugged */ -static void ad1984a_touchsmart_automic(struct hda_codec *codec) -{ - if (snd_hda_jack_detect(codec, 0x1c)) - snd_hda_codec_write(codec, 0x0c, 0, - AC_VERB_SET_CONNECT_SEL, 0x4); - else - snd_hda_codec_write(codec, 0x0c, 0, - AC_VERB_SET_CONNECT_SEL, 0x5); -} - - -/* unsolicited event for HP jack sensing */ -static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case AD1884A_HP_EVENT: - ad1884a_hp_automute(codec); - break; - case AD1884A_MIC_EVENT: - ad1984a_touchsmart_automic(codec); - break; - } -} - -/* initialize jack-sensing, too */ -static int ad1984a_touchsmart_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1884a_hp_automute(codec); - ad1984a_touchsmart_automic(codec); - return 0; -} - - -/* - */ - -enum { - AD1884A_AUTO, - AD1884A_DESKTOP, - AD1884A_LAPTOP, - AD1884A_MOBILE, - AD1884A_THINKPAD, - AD1984A_TOUCHSMART, - AD1984A_PRECISION, - AD1884A_MODELS -}; - -static const char * const ad1884a_models[AD1884A_MODELS] = { - [AD1884A_AUTO] = "auto", - [AD1884A_DESKTOP] = "desktop", - [AD1884A_LAPTOP] = "laptop", - [AD1884A_MOBILE] = "mobile", - [AD1884A_THINKPAD] = "thinkpad", - [AD1984A_TOUCHSMART] = "touchsmart", - [AD1984A_PRECISION] = "precision", -}; - -static const struct snd_pci_quirk ad1884a_cfg_tbl[] = { - SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION), - SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP), - SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), - SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), - SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE), - SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), - SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART), - {} -}; - -static int patch_ad1884a(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1884A_MODELS, - ad1884a_models, - ad1884a_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1884A_AUTO; - } - - if (board_config == AD1884A_AUTO) - return ad1884_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids); - spec->multiout.dac_nids = ad1884a_dac_nids; - spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT; - spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids); - spec->adc_nids = ad1884a_adc_nids; - spec->capsrc_nids = ad1884a_capsrc_nids; - spec->input_mux = &ad1884a_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1884a_base_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1884a_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1884a_loopbacks; -#endif - codec->patch_ops = ad198x_patch_ops; - - /* override some parameters */ - switch (board_config) { - case AD1884A_LAPTOP: - spec->mixers[0] = ad1884a_laptop_mixers; - spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs; - spec->multiout.dig_out_nid = 0; - codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event; - codec->patch_ops.init = ad1884a_laptop_init; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - case AD1884A_MOBILE: - spec->mixers[0] = ad1884a_mobile_mixers; - spec->init_verbs[0] = ad1884a_mobile_verbs; - spec->multiout.dig_out_nid = 0; - codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; - codec->patch_ops.init = ad1884a_hp_init; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - case AD1884A_THINKPAD: - spec->mixers[0] = ad1984a_thinkpad_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1984a_thinkpad_verbs; - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1984a_thinkpad_capture_source; - codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event; - codec->patch_ops.init = ad1984a_thinkpad_init; - break; - case AD1984A_PRECISION: - spec->mixers[0] = ad1984a_precision_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1984a_precision_verbs; - spec->multiout.dig_out_nid = 0; - codec->patch_ops.unsol_event = ad1984a_precision_unsol_event; - codec->patch_ops.init = ad1984a_precision_init; - break; - case AD1984A_TOUCHSMART: - spec->mixers[0] = ad1984a_touchsmart_mixers; - spec->init_verbs[0] = ad1984a_touchsmart_verbs; - spec->multiout.dig_out_nid = 0; - codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event; - codec->patch_ops.init = ad1984a_touchsmart_init; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - } - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1884a ad1884_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - /* * AD1882 / AD1882A * @@ -4923,15 +3611,15 @@ static int patch_ad1882(struct hda_codec *codec) * patch entries */ static const struct hda_codec_preset snd_hda_preset_analog[] = { - { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a }, + { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 }, { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, - { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a }, + { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, - { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a }, - { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a }, + { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 }, + { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, - { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, + { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, -- cgit v1.2.3 From bd450dcc357646cc277c560ab24b35f940efa585 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 15:48:04 +0200 Subject: ALSA: hda - Remove static quirks for AD1981 and AD1983 codecs These are relatively easy ones, as we already converted all static quirks to the generic parser. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 684 +------------------------------------------ 1 file changed, 2 insertions(+), 682 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index bfa8f532841d..4fedd9dfd85a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1427,161 +1427,6 @@ static int patch_ad1986a(struct hda_codec *codec) * AD1983 specific */ -#ifdef ENABLE_AD_STATIC_QUIRKS -#define AD1983_SPDIF_OUT 0x02 -#define AD1983_DAC 0x03 -#define AD1983_ADC 0x04 - -static const hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC }; -static const hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC }; -static const hda_nid_t ad1983_capsrc_nids[1] = { 0x15 }; - -static const struct hda_input_mux ad1983_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Line", 0x1 }, - { "Mix", 0x2 }, - { "Mix Mono", 0x3 }, - }, -}; - -/* - * SPDIF playback route - */ -static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { "PCM", "ADC" }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 2; - if (uinfo->value.enumerated.item > 1) - uinfo->value.enumerated.item = 1; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int ad1983_spdif_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - - ucontrol->value.enumerated.item[0] = spec->spdif_route; - return 0; -} - -static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - - if (ucontrol->value.enumerated.item[0] > 1) - return -EINVAL; - if (spec->spdif_route != ucontrol->value.enumerated.item[0]) { - spec->spdif_route = ucontrol->value.enumerated.item[0]; - snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0, - AC_VERB_SET_CONNECT_SEL, - spec->spdif_route); - return 1; - } - return 0; -} - -static const struct snd_kcontrol_new ad1983_mixers[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct hda_verb ad1983_init_verbs[] = { - /* Front, HP, Mono; mute as default */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Beep, PCM, Mic, Line-In: mute */ - {0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Front, HP selectors; from Mix */ - {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x06, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* Mono selector; from Mix */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03}, - /* Mic selector; Mic */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Line-in selector: Line-in */ - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mic boost: 0dB */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - /* Record selector: mic */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* SPDIF route: PCM */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Front Pin */ - {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* HP Pin */ - {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, - /* Mono Pin */ - {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* Mic Pin */ - {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* Line Pin */ - {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1983_loopbacks[] = { - { 0x12, HDA_OUTPUT, 0 }, /* Mic */ - { 0x13, HDA_OUTPUT, 0 }, /* Line */ - { } /* end */ -}; -#endif - -/* models */ -enum { - AD1983_AUTO, - AD1983_BASIC, - AD1983_MODELS -}; - -static const char * const ad1983_models[AD1983_MODELS] = { - [AD1983_AUTO] = "auto", - [AD1983_BASIC] = "basic", -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - /* * SPDIF mux control for AD1983 auto-parser */ @@ -1656,7 +1501,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec) return 0; } -static int ad1983_parse_auto_config(struct hda_codec *codec) +static int patch_ad1983(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -1681,432 +1526,11 @@ static int ad1983_parse_auto_config(struct hda_codec *codec) return err; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1983(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int board_config; - int err; - - board_config = snd_hda_check_board_config(codec, AD1983_MODELS, - ad1983_models, NULL); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1983_AUTO; - } - - if (board_config == AD1983_AUTO) - return ad1983_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids); - spec->multiout.dac_nids = ad1983_dac_nids; - spec->multiout.dig_out_nid = AD1983_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = ad1983_adc_nids; - spec->capsrc_nids = ad1983_capsrc_nids; - spec->input_mux = &ad1983_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1983_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1983_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1983_loopbacks; -#endif - spec->vmaster_nid = 0x05; - - codec->patch_ops = ad198x_patch_ops; - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1983 ad1983_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * AD1981 HD specific */ -#ifdef ENABLE_AD_STATIC_QUIRKS -#define AD1981_SPDIF_OUT 0x02 -#define AD1981_DAC 0x03 -#define AD1981_ADC 0x04 - -static const hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC }; -static const hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC }; -static const hda_nid_t ad1981_capsrc_nids[1] = { 0x15 }; - -/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */ -static const struct hda_input_mux ad1981_capture_source = { - .num_items = 7, - .items = { - { "Front Mic", 0x0 }, - { "Line", 0x1 }, - { "Mix", 0x2 }, - { "Mix Mono", 0x3 }, - { "CD", 0x4 }, - { "Mic", 0x6 }, - { "Aux", 0x7 }, - }, -}; - -static const struct snd_kcontrol_new ad1981_mixers[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Aux Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Aux Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* identical with AD1983 */ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct hda_verb ad1981_init_verbs[] = { - /* Front, HP, Mono; mute as default */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Beep, PCM, Front Mic, Line, Rear Mic, Aux, CD-In: mute */ - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Front, HP selectors; from Mix */ - {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, - {0x06, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* Mono selector; from Mix */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x03}, - /* Mic Mixer; select Front Mic */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Mic boost: 0dB */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* Record selector: Front mic */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* SPDIF route: PCM */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Front Pin */ - {0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* HP Pin */ - {0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, - /* Mono Pin */ - {0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* Front & Rear Mic Pins */ - {0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* Line Pin */ - {0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - /* Digital Beep */ - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, - /* Line-Out as Input: disabled */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - { } /* end */ -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1981_loopbacks[] = { - { 0x12, HDA_OUTPUT, 0 }, /* Front Mic */ - { 0x13, HDA_OUTPUT, 0 }, /* Line */ - { 0x1b, HDA_OUTPUT, 0 }, /* Aux */ - { 0x1c, HDA_OUTPUT, 0 }, /* Mic */ - { 0x1d, HDA_OUTPUT, 0 }, /* CD */ - { } /* end */ -}; -#endif - -/* - * Patch for HP nx6320 - * - * nx6320 uses EAPD in the reverse way - EAPD-on means the internal - * speaker output enabled _and_ mute-LED off. - */ - -#define AD1981_HP_EVENT 0x37 -#define AD1981_MIC_EVENT 0x38 - -static const struct hda_verb ad1981_hp_init_verbs[] = { - {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */ - /* pin sensing on HP and Mic jacks */ - {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT}, - {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT}, - {} -}; - -/* turn on/off EAPD (+ mute HP) as a master switch */ -static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - - if (! ad198x_eapd_put(kcontrol, ucontrol)) - return 0; - /* change speaker pin appropriately */ - snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0); - /* toggle HP mute appropriately */ - snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0, - HDA_AMP_MUTE, - spec->cur_eapd ? 0 : HDA_AMP_MUTE); - return 1; -} - -/* bind volumes of both NID 0x05 and 0x06 */ -static const struct hda_bind_ctls ad1981_hp_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -/* mute internal speaker if HP is plugged */ -static void ad1981_hp_automute(struct hda_codec *codec) -{ - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x06); - snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0, - HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); -} - -/* toggle input of built-in and mic jack appropriately */ -static void ad1981_hp_automic(struct hda_codec *codec) -{ - static const struct hda_verb mic_jack_on[] = { - {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - static const struct hda_verb mic_jack_off[] = { - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {} - }; - unsigned int present; - - present = snd_hda_jack_detect(codec, 0x08); - if (present) - snd_hda_sequence_write(codec, mic_jack_on); - else - snd_hda_sequence_write(codec, mic_jack_off); -} - -/* unsolicited event for HP jack sensing */ -static void ad1981_hp_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - res >>= 26; - switch (res) { - case AD1981_HP_EVENT: - ad1981_hp_automute(codec); - break; - case AD1981_MIC_EVENT: - ad1981_hp_automic(codec); - break; - } -} - -static const struct hda_input_mux ad1981_hp_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Dock Mic", 0x1 }, - { "Mix", 0x2 }, - }, -}; - -static const struct snd_kcontrol_new ad1981_hp_mixers[] = { - HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_NID_FLAG | 0x05, - .name = "Master Playback Switch", - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad1981_hp_master_sw_put, - .private_value = 0x05, - }, - HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), -#if 0 - /* FIXME: analog mic/line loopback doesn't work with my tests... - * (although recording is OK) - */ - HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), - /* FIXME: does this laptop have analog CD connection? */ - HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), -#endif - HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x18, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -/* initialize jack-sensing, too */ -static int ad1981_hp_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1981_hp_automute(codec); - ad1981_hp_automic(codec); - return 0; -} - -/* configuration for Toshiba Laptops */ -static const struct hda_verb ad1981_toshiba_init_verbs[] = { - {0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */ - /* pin sensing on HP and Mic jacks */ - {0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT}, - {0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT}, - {} -}; - -static const struct snd_kcontrol_new ad1981_toshiba_mixers[] = { - HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT), - { } -}; - -/* configuration for Lenovo Thinkpad T60 */ -static const struct snd_kcontrol_new ad1981_thinkpad_mixers[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - /* identical with AD1983 */ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = ad1983_spdif_route_info, - .get = ad1983_spdif_route_get, - .put = ad1983_spdif_route_put, - }, - { } /* end */ -}; - -static const struct hda_input_mux ad1981_thinkpad_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Mix", 0x2 }, - { "CD", 0x4 }, - }, -}; - -/* models */ -enum { - AD1981_AUTO, - AD1981_BASIC, - AD1981_HP, - AD1981_THINKPAD, - AD1981_TOSHIBA, - AD1981_MODELS -}; - -static const char * const ad1981_models[AD1981_MODELS] = { - [AD1981_AUTO] = "auto", - [AD1981_HP] = "hp", - [AD1981_THINKPAD] = "thinkpad", - [AD1981_BASIC] = "basic", - [AD1981_TOSHIBA] = "toshiba" -}; - -static const struct snd_pci_quirk ad1981_cfg_tbl[] = { - SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), - SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), - /* All HP models */ - SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP), - SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), - /* Lenovo Thinkpad T60/X60/Z6xx */ - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD), - /* HP nx6320 (reversed SSID, H/W bug) */ - SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP), - {} -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ - - /* follow EAPD via vmaster hook */ static void ad_vmaster_eapd_hook(void *private_data, int enabled) { @@ -2172,7 +1596,7 @@ static const struct snd_pci_quirk ad1981_fixup_tbl[] = { {} }; -static int ad1981_parse_auto_config(struct hda_codec *codec) +static int patch_ad1981(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -2205,110 +1629,6 @@ static int ad1981_parse_auto_config(struct hda_codec *codec) return err; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1981(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1981_MODELS, - ad1981_models, - ad1981_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1981_AUTO; - } - - if (board_config == AD1981_AUTO) - return ad1981_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return -ENOMEM; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids); - spec->multiout.dac_nids = ad1981_dac_nids; - spec->multiout.dig_out_nid = AD1981_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = ad1981_adc_nids; - spec->capsrc_nids = ad1981_capsrc_nids; - spec->input_mux = &ad1981_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1981_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1981_init_verbs; - spec->spdif_route = 0; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1981_loopbacks; -#endif - spec->vmaster_nid = 0x05; - - codec->patch_ops = ad198x_patch_ops; - - /* override some parameters */ - switch (board_config) { - case AD1981_HP: - spec->mixers[0] = ad1981_hp_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1981_hp_init_verbs; - if (!is_jack_available(codec, 0x0a)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1981_hp_capture_source; - - codec->patch_ops.init = ad1981_hp_init; - codec->patch_ops.unsol_event = ad1981_hp_unsol_event; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - case AD1981_THINKPAD: - spec->mixers[0] = ad1981_thinkpad_mixers; - spec->input_mux = &ad1981_thinkpad_capture_source; - /* set the upper-limit for mixer amp to 0dB for avoiding the - * possible damage by overloading - */ - snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT, - (0x17 << AC_AMPCAP_OFFSET_SHIFT) | - (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (1 << AC_AMPCAP_MUTE_SHIFT)); - break; - case AD1981_TOSHIBA: - spec->mixers[0] = ad1981_hp_mixers; - spec->mixers[1] = ad1981_toshiba_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1981_toshiba_init_verbs; - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1981_hp_capture_source; - codec->patch_ops.init = ad1981_hp_init; - codec->patch_ops.unsol_event = ad1981_hp_unsol_event; - break; - } - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1981 ad1981_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * AD1988 -- cgit v1.2.3 From 36ad45309be840d652394cfb032b592b6a20a3dd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 16:34:20 +0200 Subject: ALSA: hda - Remove static quirks for AD1988 codecs For removing static quirks for AD1988 variants, a new fixup defining the 6stack pinconfig has been added for the buggy BIOS. Other than that, we can cut off straightforwardly. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 845 ++----------------------------------------- 1 file changed, 35 insertions(+), 810 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 4fedd9dfd85a..7777a3a5f59a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1715,90 +1715,7 @@ static int patch_ad1981(struct hda_codec *codec) * E/F quad mic array */ - #ifdef ENABLE_AD_STATIC_QUIRKS -/* models */ -enum { - AD1988_AUTO, - AD1988_6STACK, - AD1988_6STACK_DIG, - AD1988_3STACK, - AD1988_3STACK_DIG, - AD1988_LAPTOP, - AD1988_LAPTOP_DIG, - AD1988_MODEL_LAST, -}; - -/* reivision id to check workarounds */ -#define AD1988A_REV2 0x100200 - -#define is_rev2(codec) \ - ((codec)->vendor_id == 0x11d41988 && \ - (codec)->revision_id == AD1988A_REV2) - -/* - * mixers - */ - -static const hda_nid_t ad1988_6stack_dac_nids[4] = { - 0x04, 0x06, 0x05, 0x0a -}; - -static const hda_nid_t ad1988_3stack_dac_nids[3] = { - 0x04, 0x05, 0x0a -}; - -/* for AD1988A revision-2, DAC2-4 are swapped */ -static const hda_nid_t ad1988_6stack_dac_nids_rev2[4] = { - 0x04, 0x05, 0x0a, 0x06 -}; - -static const hda_nid_t ad1988_alt_dac_nid[1] = { - 0x03 -}; - -static const hda_nid_t ad1988_3stack_dac_nids_rev2[3] = { - 0x04, 0x0a, 0x06 -}; - -static const hda_nid_t ad1988_adc_nids[3] = { - 0x08, 0x09, 0x0f -}; - -static const hda_nid_t ad1988_capsrc_nids[3] = { - 0x0c, 0x0d, 0x0e -}; - -#define AD1988_SPDIF_OUT 0x02 -#define AD1988_SPDIF_OUT_HDMI 0x0b -#define AD1988_SPDIF_IN 0x07 - -static const hda_nid_t ad1989b_slave_dig_outs[] = { - AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0 -}; - -static const struct hda_input_mux ad1988_6stack_capture_source = { - .num_items = 5, - .items = { - { "Front Mic", 0x1 }, /* port-B */ - { "Line", 0x2 }, /* port-C */ - { "Mic", 0x4 }, /* port-E */ - { "CD", 0x5 }, - { "Mix", 0x9 }, - }, -}; - -static const struct hda_input_mux ad1988_laptop_capture_source = { - .num_items = 3, - .items = { - { "Mic/Line", 0x1 }, /* port-B */ - { "CD", 0x5 }, - { "Mix", 0x9 }, - }, -}; - -/* - */ static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -1829,569 +1746,6 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, spec->multiout.num_dacs = spec->multiout.max_channels / 2; return err; } - -/* 6-stack mode */ -static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Side Playback Volume", 0x06, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT), - HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), - HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -/* 3-stack mode */ -static const struct snd_kcontrol_new ad1988_3stack_mixers1[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x06, 2, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT), - HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT), - HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x26, 2, 2, HDA_INPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT), - HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = ad198x_ch_mode_info, - .get = ad198x_ch_mode_get, - .put = ad198x_ch_mode_put, - }, - - { } /* end */ -}; - -/* laptop mode */ -static const struct snd_kcontrol_new ad1988_laptop_mixers[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT), - HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT), - - HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT), - - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "External Amplifier", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x12, - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad198x_eapd_put, - .private_value = 0x12, /* port-D */ - }, - - { } /* end */ -}; - -/* capture */ -static const struct snd_kcontrol_new ad1988_capture_mixers[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x0e, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x0e, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 3, - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - static const char * const texts[] = { - "PCM", "ADC1", "ADC2", "ADC3" - }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item >= 4) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int sel; - - sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_INPUT); - if (!(sel & 0x80)) - ucontrol->value.enumerated.item[0] = 0; - else { - sel = snd_hda_codec_read(codec, 0x0b, 0, - AC_VERB_GET_CONNECT_SEL, 0); - if (sel < 3) - sel++; - else - sel = 0; - ucontrol->value.enumerated.item[0] = sel; - } - return 0; -} - -static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - unsigned int val, sel; - int change; - - val = ucontrol->value.enumerated.item[0]; - if (val > 3) - return -EINVAL; - if (!val) { - sel = snd_hda_codec_read(codec, 0x1d, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_INPUT); - change = sel & 0x80; - if (change) { - snd_hda_codec_write_cache(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_UNMUTE(0)); - snd_hda_codec_write_cache(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_MUTE(1)); - } - } else { - sel = snd_hda_codec_read(codec, 0x1d, 0, - AC_VERB_GET_AMP_GAIN_MUTE, - AC_AMP_GET_INPUT | 0x01); - change = sel & 0x80; - if (change) { - snd_hda_codec_write_cache(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_MUTE(0)); - snd_hda_codec_write_cache(codec, 0x1d, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_UNMUTE(1)); - } - sel = snd_hda_codec_read(codec, 0x0b, 0, - AC_VERB_GET_CONNECT_SEL, 0) + 1; - change |= sel != val; - if (change) - snd_hda_codec_write_cache(codec, 0x0b, 0, - AC_VERB_SET_CONNECT_SEL, - val - 1); - } - return change; -} - -static const struct snd_kcontrol_new ad1988_spdif_out_mixers[] = { - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "IEC958 Playback Source", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, - .info = ad1988_spdif_playback_source_info, - .get = ad1988_spdif_playback_source_get, - .put = ad1988_spdif_playback_source_put, - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { - HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { - HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -/* - * initialization verbs - */ - -/* - * for 6-stack (+dig) - */ -static const struct hda_verb ad1988_6stack_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; unmute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Port-D line-out path */ - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-F surround path */ - {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-G CLFE path */ - {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Port-H side path */ - {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mono out path */ - {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ - /* Port-B front mic-in path */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-C line-in path */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Port-E mic-in path */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Analog CD Input */ - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - - { } -}; - -static const struct hda_verb ad1988_6stack_fp_init_verbs[] = { - /* Headphone; unmute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - - { } -}; - -static const struct hda_verb ad1988_capture_init_verbs[] = { - /* mute analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* select ADCs - front-mic */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - - { } -}; - -static const struct hda_verb ad1988_spdif_init_verbs[] = { - /* SPDIF out sel */ - {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */ - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* SPDIF out pin */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - - { } -}; - -static const struct hda_verb ad1988_spdif_in_init_verbs[] = { - /* unmute SPDIF input pin */ - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { } -}; - -/* AD1989 has no ADC -> SPDIF route */ -static const struct hda_verb ad1989_spdif_init_verbs[] = { - /* SPDIF-1 out pin */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - /* SPDIF-2/HDMI out pin */ - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - { } -}; - -/* - * verbs for 3stack (+dig) - */ -static const struct hda_verb ad1988_3stack_ch2_init[] = { - /* set port-C to line-in */ - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - /* set port-E to mic-in */ - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { } /* end */ -}; - -static const struct hda_verb ad1988_3stack_ch6_init[] = { - /* set port-C to surround out */ - { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - /* set port-E to CLFE out */ - { 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } /* end */ -}; - -static const struct hda_channel_mode ad1988_3stack_modes[2] = { - { 2, ad1988_3stack_ch2_init }, - { 6, ad1988_3stack_ch6_init }, -}; - -static const struct hda_verb ad1988_3stack_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; unmute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* Port-D line-out path */ - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - /* Mono out path */ - {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ - /* Port-B front mic-in path */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-C line-in/surround path - 6ch mode as default */ - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x31, AC_VERB_SET_CONNECT_SEL, 0x0}, /* output sel: DAC 0x05 */ - {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Port-E mic-in/CLFE path - 6ch mode as default */ - {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x0a */ - {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* mute analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* select ADCs - front-mic */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - { } -}; - -/* - * verbs for laptop mode (+dig) - */ -static const struct hda_verb ad1988_laptop_hp_on[] = { - /* unmute port-A and mute port-D */ - { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } /* end */ -}; -static const struct hda_verb ad1988_laptop_hp_off[] = { - /* mute port-A and unmute port-D */ - { 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } /* end */ -}; - -#define AD1988_HP_EVENT 0x01 - -static const struct hda_verb ad1988_laptop_init_verbs[] = { - /* Front, Surround, CLFE, side DAC; unmute as default */ - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Port-A front headphon path */ - {0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */ - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - /* unsolicited event for pin-sense */ - {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1988_HP_EVENT }, - /* Port-D line-out path + EAPD */ - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, /* EAPD-off */ - /* Mono out path */ - {0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */ - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */ - /* Port-B mic-in path */ - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - /* Port-C docking station - try to output */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x33, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* mute analog mix */ - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, - /* select ADCs - mic */ - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Analog Mix output amp */ - {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ - { } -}; - -static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) != AD1988_HP_EVENT) - return; - if (snd_hda_jack_detect(codec, 0x11)) - snd_hda_sequence_write(codec, ad1988_laptop_hp_on); - else - snd_hda_sequence_write(codec, ad1988_laptop_hp_off); -} - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1988_loopbacks[] = { - { 0x20, HDA_INPUT, 0 }, /* Front Mic */ - { 0x20, HDA_INPUT, 1 }, /* Line */ - { 0x20, HDA_INPUT, 4 }, /* Mic */ - { 0x20, HDA_INPUT, 6 }, /* CD */ - { } /* end */ -}; -#endif #endif /* ENABLE_AD_STATIC_QUIRKS */ static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol, @@ -2540,7 +1894,34 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec) /* */ -static int ad1988_parse_auto_config(struct hda_codec *codec) +enum { + AD1988_FIXUP_6STACK_DIG, +}; + +static const struct hda_fixup ad1988_fixups[] = { + [AD1988_FIXUP_6STACK_DIG] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x11, 0x02214130 }, /* front-hp */ + { 0x12, 0x01014010 }, /* line-out */ + { 0x14, 0x02a19122 }, /* front-mic */ + { 0x15, 0x01813021 }, /* line-in */ + { 0x16, 0x01011012 }, /* line-out */ + { 0x17, 0x01a19020 }, /* mic */ + { 0x1b, 0x0145f1f0 }, /* SPDIF */ + { 0x24, 0x01016011 }, /* line-out */ + { 0x25, 0x01012013 }, /* line-out */ + { } + } + }, +}; + +static const struct hda_model_fixup ad1988_fixup_models[] = { + { .id = AD1988_FIXUP_6STACK_DIG, .name = "6stack-dig" }, + {} +}; + +static int patch_ad1988(struct hda_codec *codec) { struct ad198x_spec *spec; int err; @@ -2554,12 +1935,19 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) spec->gen.mixer_merge_nid = 0x21; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + + snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups); + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + err = ad198x_parse_auto_config(codec); if (err < 0) goto error; err = ad1988_add_spdif_mux_ctl(codec); if (err < 0) goto error; + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; error: @@ -2567,169 +1955,6 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) return err; } -/* - */ - -#ifdef ENABLE_AD_STATIC_QUIRKS -static const char * const ad1988_models[AD1988_MODEL_LAST] = { - [AD1988_6STACK] = "6stack", - [AD1988_6STACK_DIG] = "6stack-dig", - [AD1988_3STACK] = "3stack", - [AD1988_3STACK_DIG] = "3stack-dig", - [AD1988_LAPTOP] = "laptop", - [AD1988_LAPTOP_DIG] = "laptop-dig", - [AD1988_AUTO] = "auto", -}; - -static const struct snd_pci_quirk ad1988_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), - SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), - SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), - SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG), - SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG), - {} -}; - -static int patch_ad1988(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST, - ad1988_models, ad1988_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1988_AUTO; - } - - if (board_config == AD1988_AUTO) - return ad1988_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - if (is_rev2(codec)) - snd_printk(KERN_INFO "patch_analog: AD1988A rev.2 is detected, enable workarounds\n"); - - err = snd_hda_attach_beep_device(codec, 0x10); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); - - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = ad1988_alt_dac_nid[0]; - switch (board_config) { - case AD1988_6STACK: - case AD1988_6STACK_DIG: - spec->multiout.max_channels = 8; - spec->multiout.num_dacs = 4; - if (is_rev2(codec)) - spec->multiout.dac_nids = ad1988_6stack_dac_nids_rev2; - else - spec->multiout.dac_nids = ad1988_6stack_dac_nids; - spec->input_mux = &ad1988_6stack_capture_source; - spec->num_mixers = 2; - if (is_rev2(codec)) - spec->mixers[0] = ad1988_6stack_mixers1_rev2; - else - spec->mixers[0] = ad1988_6stack_mixers1; - spec->mixers[1] = ad1988_6stack_mixers2; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1988_6stack_init_verbs; - if (board_config == AD1988_6STACK_DIG) { - spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; - spec->dig_in_nid = AD1988_SPDIF_IN; - } - break; - case AD1988_3STACK: - case AD1988_3STACK_DIG: - spec->multiout.max_channels = 6; - spec->multiout.num_dacs = 3; - if (is_rev2(codec)) - spec->multiout.dac_nids = ad1988_3stack_dac_nids_rev2; - else - spec->multiout.dac_nids = ad1988_3stack_dac_nids; - spec->input_mux = &ad1988_6stack_capture_source; - spec->channel_mode = ad1988_3stack_modes; - spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes); - spec->num_mixers = 2; - if (is_rev2(codec)) - spec->mixers[0] = ad1988_3stack_mixers1_rev2; - else - spec->mixers[0] = ad1988_3stack_mixers1; - spec->mixers[1] = ad1988_3stack_mixers2; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1988_3stack_init_verbs; - if (board_config == AD1988_3STACK_DIG) - spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; - break; - case AD1988_LAPTOP: - case AD1988_LAPTOP_DIG: - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1988_3stack_dac_nids; - spec->input_mux = &ad1988_laptop_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1988_laptop_mixers; - codec->inv_eapd = 1; /* inverted EAPD */ - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1988_laptop_init_verbs; - if (board_config == AD1988_LAPTOP_DIG) - spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; - break; - } - - spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids); - spec->adc_nids = ad1988_adc_nids; - spec->capsrc_nids = ad1988_capsrc_nids; - spec->mixers[spec->num_mixers++] = ad1988_capture_mixers; - spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs; - if (spec->multiout.dig_out_nid) { - if (codec->vendor_id >= 0x11d4989a) { - spec->mixers[spec->num_mixers++] = - ad1989_spdif_out_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1989_spdif_init_verbs; - codec->slave_dig_outs = ad1989b_slave_dig_outs; - } else { - spec->mixers[spec->num_mixers++] = - ad1988_spdif_out_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1988_spdif_init_verbs; - } - } - if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) { - spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers; - spec->init_verbs[spec->num_init_verbs++] = - ad1988_spdif_in_init_verbs; - } - - codec->patch_ops = ad198x_patch_ops; - switch (board_config) { - case AD1988_LAPTOP: - case AD1988_LAPTOP_DIG: - codec->patch_ops.unsol_event = ad1988_laptop_unsol_event; - break; - } -#ifdef CONFIG_PM - spec->loopback.amplist = ad1988_loopbacks; -#endif - spec->vmaster_nid = 0x04; - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1988 ad1988_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * AD1884 / AD1984 -- cgit v1.2.3 From e0b27167c2d6464ff7ae7e35725024349e44596b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 16:50:46 +0200 Subject: ALSA: hda - Convert the static quirk for Samsung Q1 Ultra ... to a fixup entry. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 33 ++++++++++----------------------- 1 file changed, 10 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 7777a3a5f59a..056810c14e71 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1063,17 +1063,6 @@ static const struct hda_verb ad1986a_automic_verbs[] = { {} }; -/* Ultra initialization */ -static const struct hda_verb ad1986a_ultra_init[] = { - /* eapd initialization */ - { 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, - /* CLFE -> Mic in */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2 }, - { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - { 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 }, - { } /* end */ -}; - /* pin sensing on HP jack */ static const struct hda_verb ad1986a_hp_init_verbs[] = { {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT}, @@ -1110,7 +1099,6 @@ enum { AD1986A_LAPTOP, AD1986A_LAPTOP_EAPD, AD1986A_LAPTOP_AUTOMUTE, - AD1986A_ULTRA, AD1986A_SAMSUNG, AD1986A_SAMSUNG_P50, AD1986A_MODELS @@ -1123,7 +1111,6 @@ static const char * const ad1986a_models[AD1986A_MODELS] = { [AD1986A_LAPTOP] = "laptop", [AD1986A_LAPTOP_EAPD] = "laptop-eapd", [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute", - [AD1986A_ULTRA] = "ultra", [AD1986A_SAMSUNG] = "samsung", [AD1986A_SAMSUNG_P50] = "samsung-p50", }; @@ -1149,7 +1136,6 @@ static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50), - SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), @@ -1203,6 +1189,7 @@ static void ad_fixup_inv_jack_detect(struct hda_codec *codec, enum { AD1986A_FIXUP_INV_JACK_DETECT, + AD1986A_FIXUP_ULTRA, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -1210,9 +1197,18 @@ static const struct hda_fixup ad1986a_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = ad_fixup_inv_jack_detect, }, + [AD1986A_FIXUP_ULTRA] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1d, 0x90a7013e }, /* int mic */ + {} + }, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { + SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT), {} }; @@ -1395,15 +1391,6 @@ static int patch_ad1986a(struct hda_codec *codec) */ spec->inv_jack_detect = 1; break; - case AD1986A_ULTRA: - spec->mixers[0] = ad1986a_laptop_eapd_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1986a_ultra_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - spec->multiout.dig_out_nid = 0; - break; } /* AD1986A has a hardware problem that it can't share a stream -- cgit v1.2.3 From f8c0ab1798b601493f29cb4836ccdaa3811ba390 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 17:06:04 +0200 Subject: ALSA: hda - Convert static quirks for AD1986A Samsung laptops Just need to override some pin-configurations. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 108 +++++-------------------------------------- 1 file changed, 12 insertions(+), 96 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 056810c14e71..1e4dc98e321a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -857,33 +857,6 @@ static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = { { } /* end */ }; -/* re-connect the mic boost input according to the jack sensing */ -static void ad1986a_automic(struct hda_codec *codec) -{ - unsigned int present; - present = snd_hda_jack_detect(codec, 0x1f); - /* 0 = 0x1f, 2 = 0x1d, 4 = mixed */ - snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL, - present ? 0 : 2); -} - -#define AD1986A_MIC_EVENT 0x36 - -static void ad1986a_automic_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - if ((res >> 26) != AD1986A_MIC_EVENT) - return; - ad1986a_automic(codec); -} - -static int ad1986a_automic_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1986a_automic(codec); - return 0; -} - /* laptop-automute - 2ch only */ static void ad1986a_update_hp(struct hda_codec *codec) @@ -1054,42 +1027,12 @@ static const struct hda_verb ad1986a_eapd_init_verbs[] = { {} }; -static const struct hda_verb ad1986a_automic_verbs[] = { - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - /*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/ - {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT}, - {} -}; - /* pin sensing on HP jack */ static const struct hda_verb ad1986a_hp_init_verbs[] = { {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT}, {} }; -static void ad1986a_samsung_p50_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - switch (res >> 26) { - case AD1986A_HP_EVENT: - ad1986a_hp_automute(codec); - break; - case AD1986A_MIC_EVENT: - ad1986a_automic(codec); - break; - } -} - -static int ad1986a_samsung_p50_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1986a_hp_automute(codec); - ad1986a_automic(codec); - return 0; -} - /* models */ enum { @@ -1099,8 +1042,6 @@ enum { AD1986A_LAPTOP, AD1986A_LAPTOP_EAPD, AD1986A_LAPTOP_AUTOMUTE, - AD1986A_SAMSUNG, - AD1986A_SAMSUNG_P50, AD1986A_MODELS }; @@ -1111,8 +1052,6 @@ static const char * const ad1986a_models[AD1986A_MODELS] = { [AD1986A_LAPTOP] = "laptop", [AD1986A_LAPTOP_EAPD] = "laptop-eapd", [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute", - [AD1986A_SAMSUNG] = "samsung", - [AD1986A_SAMSUNG_P50] = "samsung-p50", }; static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { @@ -1135,8 +1074,6 @@ static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50), - SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), @@ -1190,6 +1127,7 @@ static void ad_fixup_inv_jack_detect(struct hda_codec *codec, enum { AD1986A_FIXUP_INV_JACK_DETECT, AD1986A_FIXUP_ULTRA, + AD1986A_FIXUP_SAMSUNG, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -1205,9 +1143,20 @@ static const struct hda_fixup ad1986a_fixups[] = { {} }, }, + [AD1986A_FIXUP_SAMSUNG] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1d, 0x90a7013e }, /* int mic */ + { 0x20, 0x411111f0 }, /* N/A */ + { 0x24, 0x411111f0 }, /* N/A */ + {} + }, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT), {} @@ -1337,39 +1286,6 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.dig_out_nid = 0; spec->input_mux = &ad1986a_laptop_eapd_capture_source; break; - case AD1986A_SAMSUNG: - spec->num_mixers = 2; - spec->mixers[0] = ad1986a_laptop_master_mixers; - spec->mixers[1] = ad1986a_laptop_eapd_mixers; - spec->num_init_verbs = 3; - spec->init_verbs[1] = ad1986a_eapd_init_verbs; - spec->init_verbs[2] = ad1986a_automic_verbs; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - if (!is_jack_available(codec, 0x25)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_automic_capture_source; - codec->patch_ops.unsol_event = ad1986a_automic_unsol_event; - codec->patch_ops.init = ad1986a_automic_init; - break; - case AD1986A_SAMSUNG_P50: - spec->num_mixers = 2; - spec->mixers[0] = ad1986a_automute_master_mixers; - spec->mixers[1] = ad1986a_laptop_eapd_mixers; - spec->num_init_verbs = 4; - spec->init_verbs[1] = ad1986a_eapd_init_verbs; - spec->init_verbs[2] = ad1986a_automic_verbs; - spec->init_verbs[3] = ad1986a_hp_init_verbs; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - if (!is_jack_available(codec, 0x25)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_automic_capture_source; - codec->patch_ops.unsol_event = ad1986a_samsung_p50_unsol_event; - codec->patch_ops.init = ad1986a_samsung_p50_init; - break; case AD1986A_LAPTOP_AUTOMUTE: spec->num_mixers = 3; spec->mixers[0] = ad1986a_automute_master_mixers; -- cgit v1.2.3 From 7fc116ec27cf51831d2d4e555c89d899be410340 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 17:18:48 +0200 Subject: ALSA: hda - Drop static quirks for other AD1986A Samsung machines BIOS on Samsung R55, M55 and M50 provide the proper pin-configs, so we can remove the corresponding static quirk entries gracefully. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1e4dc98e321a..3f2434ad7ce7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1072,13 +1072,10 @@ static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK), - SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), - SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP), {} }; -- cgit v1.2.3 From fc39a7ea9235104b06ee43385d4265f2d078e62b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 17:25:03 +0200 Subject: ALSA: hda - Drop static quirk for Toshiba Satellite L40-10Q The BIOS provides good pin-configurations, so we can drop the static quirk now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 3f2434ad7ce7..a41e121fe056 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1071,7 +1071,6 @@ static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), - SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), -- cgit v1.2.3 From 0f7dbda0ec3bc4d778d7acf741b220fbf4318a20 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 4 Jul 2013 18:03:56 +0200 Subject: ALSA: hda - Drop a few other static quirks for AD1986A Most of ASUS laptops and Lenovo N100 provide proper BIOS pin-configs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index a41e121fe056..3b23280ff3a6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1056,15 +1056,6 @@ static const char * const ad1986a_models[AD1986A_MODELS] = { static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1263, "ASUS U5F", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1302, "ASUS W3j", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x1443, "ASUS VX1", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8J", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP), SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK), @@ -1074,7 +1065,6 @@ static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), - SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), {} }; -- cgit v1.2.3 From 632408adfe70be6706cb89522b0d5b3dce188d84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 5 Jul 2013 14:14:14 +0200 Subject: ALSA: hda - Remove static quirks for AD1986A codec Finally all the static quirks in patch_analog.c are reduced by this patch. As machines with AD1986A codec are all old and often their BIOS are buggy, we need to keep at least a few static pin conifgs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1069 +++--------------------------------------- 1 file changed, 57 insertions(+), 1012 deletions(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 3b23280ff3a6..0cbdd87dde6d 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -32,7 +32,6 @@ #include "hda_jack.h" #include "hda_generic.h" -#define ENABLE_AD_STATIC_QUIRKS struct ad198x_spec { struct hda_gen_spec gen; @@ -43,114 +42,8 @@ struct ad198x_spec { hda_nid_t eapd_nid; unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - -#ifdef ENABLE_AD_STATIC_QUIRKS - const struct snd_kcontrol_new *mixers[6]; - int num_mixers; - const struct hda_verb *init_verbs[6]; /* initialization verbs - * don't forget NULL termination! - */ - unsigned int num_init_verbs; - - /* playback */ - struct hda_multi_out multiout; /* playback set-up - * max_channels, dacs must be set - * dig_out_nid and hp_nid are optional - */ - unsigned int cur_eapd; - unsigned int need_dac_fix; - - /* capture */ - unsigned int num_adc_nids; - const hda_nid_t *adc_nids; - hda_nid_t dig_in_nid; /* digital-in NID; optional */ - - /* capture source */ - const struct hda_input_mux *input_mux; - const hda_nid_t *capsrc_nids; - unsigned int cur_mux[3]; - - /* channel model */ - const struct hda_channel_mode *channel_mode; - int num_channel_mode; - - /* PCM information */ - struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ - - unsigned int spdif_route; - - unsigned int jack_present: 1; - unsigned int inv_jack_detect: 1;/* inverted jack-detection */ - unsigned int analog_beep: 1; /* analog beep input present */ - unsigned int avoid_init_slave_vol:1; - -#ifdef CONFIG_PM - struct hda_loopback_check loopback; -#endif - /* for virtual master */ - hda_nid_t vmaster_nid; - const char * const *slave_vols; - const char * const *slave_sws; -#endif /* ENABLE_AD_STATIC_QUIRKS */ }; -#ifdef ENABLE_AD_STATIC_QUIRKS -/* - * input MUX handling (common part) - */ -static int ad198x_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - - return snd_hda_input_mux_info(spec->input_mux, uinfo); -} - -static int ad198x_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx]; - return 0; -} - -static int ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->capsrc_nids[adc_idx], - &spec->cur_mux[adc_idx]); -} - -/* - * initialization (common callbacks) - */ -static int ad198x_init(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_init_verbs; i++) - snd_hda_sequence_write(codec, spec->init_verbs[i]); - return 0; -} - -static const char * const ad_slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", - "Headphone", "Mono", "Speaker", "IEC958", - NULL -}; - -static const char * const ad1988_6stack_fp_slave_pfxs[] = { - "Front", "Surround", "Center", "LFE", "Side", "IEC958", - NULL -}; -#endif /* ENABLE_AD_STATIC_QUIRKS */ #ifdef CONFIG_SND_HDA_INPUT_BEEP /* additional beep mixers; the actual parameters are overwritten at build */ @@ -160,12 +53,6 @@ static const struct snd_kcontrol_new ad_beep_mixer[] = { { } /* end */ }; -static const struct snd_kcontrol_new ad_beep2_mixer[] = { - HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0, 0, HDA_OUTPUT), - HDA_CODEC_MUTE_BEEP("Digital Beep Playback Switch", 0, 0, HDA_OUTPUT), - { } /* end */ -}; - #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */ #else @@ -181,8 +68,7 @@ static int create_beep_ctls(struct hda_codec *codec) if (!spec->beep_amp) return 0; - knew = spec->analog_beep ? ad_beep2_mixer : ad_beep_mixer; - for ( ; knew->name; knew++) { + for (knew = ad_beep_mixer ; knew->name; knew++) { int err; struct snd_kcontrol *kctl; kctl = snd_ctl_new1(knew, codec); @@ -199,268 +85,6 @@ static int create_beep_ctls(struct hda_codec *codec) #define create_beep_ctls(codec) 0 #endif -#ifdef ENABLE_AD_STATIC_QUIRKS -static int ad198x_build_controls(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - struct snd_kcontrol *kctl; - unsigned int i; - int err; - - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } - if (spec->multiout.dig_out_nid) { - err = snd_hda_create_spdif_out_ctls(codec, - spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); - if (err < 0) - return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; - } - if (spec->dig_in_nid) { - err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); - if (err < 0) - return err; - } - - /* create beep controls if needed */ - err = create_beep_ctls(codec); - if (err < 0) - return err; - - /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { - unsigned int vmaster_tlv[4]; - snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, vmaster_tlv); - err = __snd_hda_add_vmaster(codec, "Master Playback Volume", - vmaster_tlv, - (spec->slave_vols ? - spec->slave_vols : ad_slave_pfxs), - "Playback Volume", - !spec->avoid_init_slave_vol, NULL); - if (err < 0) - return err; - } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, - (spec->slave_sws ? - spec->slave_sws : ad_slave_pfxs), - "Playback Switch"); - if (err < 0) - return err; - } - - /* assign Capture Source enums to NID */ - kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); - if (!kctl) - kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); - for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]); - if (err < 0) - return err; - } - - /* assign IEC958 enums to NID */ - kctl = snd_hda_find_mixer_ctl(codec, - SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source"); - if (kctl) { - err = snd_hda_add_nid(codec, kctl, 0, - spec->multiout.dig_out_nid); - if (err < 0) - return err; - } - - return 0; -} - -#ifdef CONFIG_PM -static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_check_amp_list_power(codec, &spec->loopback, nid); -} -#endif - -/* - * Analog playback callbacks - */ -static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, - hinfo); -} - -static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag, - format, substream); -} - -static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); -} - -/* - * Digital out - */ -static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_dig_open(codec, &spec->multiout); -} - -static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_dig_close(codec, &spec->multiout); -} - -static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); -} - -static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); -} - -/* - * Analog capture - */ -static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - unsigned int stream_tag, - unsigned int format, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); - return 0; -} - -static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, - struct hda_codec *codec, - struct snd_pcm_substream *substream) -{ - struct ad198x_spec *spec = codec->spec; - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); - return 0; -} - -/* - */ -static const struct hda_pcm_stream ad198x_pcm_analog_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 6, /* changed later */ - .nid = 0, /* fill later */ - .ops = { - .open = ad198x_playback_pcm_open, - .prepare = ad198x_playback_pcm_prepare, - .cleanup = ad198x_playback_pcm_cleanup, - }, -}; - -static const struct hda_pcm_stream ad198x_pcm_analog_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .prepare = ad198x_capture_pcm_prepare, - .cleanup = ad198x_capture_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream ad198x_pcm_digital_playback = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - .nid = 0, /* fill later */ - .ops = { - .open = ad198x_dig_playback_pcm_open, - .close = ad198x_dig_playback_pcm_close, - .prepare = ad198x_dig_playback_pcm_prepare, - .cleanup = ad198x_dig_playback_pcm_cleanup - }, -}; - -static const struct hda_pcm_stream ad198x_pcm_digital_capture = { - .substreams = 1, - .channels_min = 2, - .channels_max = 2, - /* NID is set in alc_build_pcms */ -}; - -static int ad198x_build_pcms(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - struct hda_pcm *info = spec->pcm_rec; - - codec->num_pcms = 1; - codec->pcm_info = info; - - info->name = "AD198x Analog"; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_analog_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; - info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; - - if (spec->multiout.dig_out_nid) { - info++; - codec->num_pcms++; - codec->spdif_status_reset = 1; - info->name = "AD198x Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; - info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_digital_playback; - info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; - if (spec->dig_in_nid) { - info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_digital_capture; - info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; - } - } - - return 0; -} -#endif /* ENABLE_AD_STATIC_QUIRKS */ static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front, hda_nid_t hp) @@ -507,18 +131,6 @@ static void ad198x_shutup(struct hda_codec *codec) ad198x_power_eapd(codec); } -static void ad198x_free(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - - if (!spec) - return; - - snd_hda_gen_spec_free(&spec->gen); - kfree(spec); - snd_hda_detach_beep_device(codec); -} - #ifdef CONFIG_PM static int ad198x_suspend(struct hda_codec *codec) { @@ -527,65 +139,6 @@ static int ad198x_suspend(struct hda_codec *codec) } #endif -#ifdef ENABLE_AD_STATIC_QUIRKS -static const struct hda_codec_ops ad198x_patch_ops = { - .build_controls = ad198x_build_controls, - .build_pcms = ad198x_build_pcms, - .init = ad198x_init, - .free = ad198x_free, -#ifdef CONFIG_PM - .check_power_status = ad198x_check_power_status, - .suspend = ad198x_suspend, -#endif - .reboot_notify = ad198x_shutup, -}; - - -/* - * EAPD control - * the private value = nid - */ -#define ad198x_eapd_info snd_ctl_boolean_mono_info - -static int ad198x_eapd_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - if (codec->inv_eapd) - ucontrol->value.integer.value[0] = ! spec->cur_eapd; - else - ucontrol->value.integer.value[0] = spec->cur_eapd; - return 0; -} - -static int ad198x_eapd_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct ad198x_spec *spec = codec->spec; - hda_nid_t nid = kcontrol->private_value & 0xff; - unsigned int eapd; - eapd = !!ucontrol->value.integer.value[0]; - if (codec->inv_eapd) - eapd = !eapd; - if (eapd == spec->cur_eapd) - return 0; - spec->cur_eapd = eapd; - snd_hda_codec_write_cache(codec, nid, - 0, AC_VERB_SET_EAPD_BTLENABLE, - eapd ? 0x02 : 0x00); - return 1; -} - -static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo); -static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol); -#endif /* ENABLE_AD_STATIC_QUIRKS */ - /* * Automatic parse of I/O pins from the BIOS configuration @@ -646,446 +199,6 @@ static int ad198x_parse_auto_config(struct hda_codec *codec) * AD1986A specific */ -#ifdef ENABLE_AD_STATIC_QUIRKS -#define AD1986A_SPDIF_OUT 0x02 -#define AD1986A_FRONT_DAC 0x03 -#define AD1986A_SURR_DAC 0x04 -#define AD1986A_CLFE_DAC 0x05 -#define AD1986A_ADC 0x06 - -static const hda_nid_t ad1986a_dac_nids[3] = { - AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC -}; -static const hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC }; -static const hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 }; - -static const struct hda_input_mux ad1986a_capture_source = { - .num_items = 7, - .items = { - { "Mic", 0x0 }, - { "CD", 0x1 }, - { "Aux", 0x3 }, - { "Line", 0x4 }, - { "Mix", 0x5 }, - { "Mono", 0x6 }, - { "Phone", 0x7 }, - }, -}; - - -static const struct hda_bind_ctls ad1986a_bind_pcm_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -static const struct hda_bind_ctls ad1986a_bind_pcm_sw = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT), - 0 - }, -}; - -/* - * mixers - */ -static const struct snd_kcontrol_new ad1986a_mixers[] = { - /* - * bind volumes/mutes of 3 DACs as a single PCM control for simplicity - */ - HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol), - HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw), - HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Surround Playback Switch", 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x1d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x1d, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x1d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x1d, 2, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT), - { } /* end */ -}; - -/* additional mixers for 3stack mode */ -static const struct snd_kcontrol_new ad1986a_3st_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Channel Mode", - .info = ad198x_ch_mode_info, - .get = ad198x_ch_mode_get, - .put = ad198x_ch_mode_put, - }, - { } /* end */ -}; - -/* laptop model - 2ch only */ -static const hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC }; - -/* master controls both pins 0x1a and 0x1b */ -static const struct hda_bind_ctls ad1986a_laptop_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - 0, - }, -}; - -static const struct hda_bind_ctls ad1986a_laptop_master_sw = { - .ops = &snd_hda_bind_sw, - .values = { - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), - 0, - }, -}; - -static const struct snd_kcontrol_new ad1986a_laptop_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), - HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), - HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT), - /* - HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */ - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { } /* end */ -}; - -/* laptop-eapd model - 2ch only */ - -static const struct hda_input_mux ad1986a_laptop_eapd_capture_source = { - .num_items = 3, - .items = { - { "Mic", 0x0 }, - { "Internal Mic", 0x4 }, - { "Mix", 0x5 }, - }, -}; - -static const struct hda_input_mux ad1986a_automic_capture_source = { - .num_items = 2, - .items = { - { "Mic", 0x0 }, - { "Mix", 0x5 }, - }, -}; - -static const struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = { - HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), - HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { - HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Mic Boost Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = ad198x_mux_enum_info, - .get = ad198x_mux_enum_get, - .put = ad198x_mux_enum_put, - }, - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "External Amplifier", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad198x_eapd_put, - .private_value = 0x1b, /* port-D */ - }, - { } /* end */ -}; - -static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT), - { } /* end */ -}; - -/* laptop-automute - 2ch only */ - -static void ad1986a_update_hp(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - unsigned int mute; - - if (spec->jack_present) - mute = HDA_AMP_MUTE; /* mute internal speaker */ - else - /* unmute internal speaker if necessary */ - mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0); - snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); -} - -static void ad1986a_hp_automute(struct hda_codec *codec) -{ - struct ad198x_spec *spec = codec->spec; - - spec->jack_present = snd_hda_jack_detect(codec, 0x1a); - if (spec->inv_jack_detect) - spec->jack_present = !spec->jack_present; - ad1986a_update_hp(codec); -} - -#define AD1986A_HP_EVENT 0x37 - -static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res) -{ - if ((res >> 26) != AD1986A_HP_EVENT) - return; - ad1986a_hp_automute(codec); -} - -static int ad1986a_hp_init(struct hda_codec *codec) -{ - ad198x_init(codec); - ad1986a_hp_automute(codec); - return 0; -} - -/* bind hp and internal speaker mute (with plug check) */ -static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); - if (change) - ad1986a_update_hp(codec); - return change; -} - -static const struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { - HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_AMP_FLAG, - .info = snd_hda_mixer_amp_switch_info, - .get = snd_hda_mixer_amp_switch_get, - .put = ad1986a_hp_master_sw_put, - .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), - }, - { } /* end */ -}; - - -/* - * initialization verbs - */ -static const struct hda_verb ad1986a_init_verbs[] = { - /* Front, Surround, CLFE DAC; mute as default */ - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* Downmix - off */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* HP, Line-Out, Surround, CLFE selectors */ - {0x0a, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, - {0x0d, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mono selector */ - {0x0e, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mic selector: Mic 1/2 pin */ - {0x0f, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Line-in selector: Line-in */ - {0x10, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mic 1/2 swap */ - {0x11, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Record selector: mic */ - {0x12, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Mic, Phone, CD, Aux, Line-In amp; mute as default */ - {0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* PC beep */ - {0x18, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* HP, Line-Out, Surround, CLFE, Mono pins; mute as default */ - {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, - /* HP Pin */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 }, - /* Front, Surround, CLFE Pins */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* Mono Pin */ - {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 }, - /* Mic Pin */ - {0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 }, - /* Line, Aux, CD, Beep-In Pin */ - {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - {0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 }, - { } /* end */ -}; - -static const struct hda_verb ad1986a_ch2_init[] = { - /* Surround out -> Line In */ - { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, - /* Line-in selectors */ - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x1 }, - /* CLFE -> Mic in */ - { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - /* Mic selector, mix C/LFE (backmic) and Mic (frontmic) */ - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 }, - { } /* end */ -}; - -static const struct hda_verb ad1986a_ch4_init[] = { - /* Surround out -> Surround */ - { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* CLFE -> Mic in */ - { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 }, - { } /* end */ -}; - -static const struct hda_verb ad1986a_ch6_init[] = { - /* Surround out -> Surround out */ - { 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 }, - /* CLFE -> CLFE */ - { 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x0 }, - { } /* end */ -}; - -static const struct hda_channel_mode ad1986a_modes[3] = { - { 2, ad1986a_ch2_init }, - { 4, ad1986a_ch4_init }, - { 6, ad1986a_ch6_init }, -}; - -/* eapd initialization */ -static const struct hda_verb ad1986a_eapd_init_verbs[] = { - {0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, - {} -}; - -/* pin sensing on HP jack */ -static const struct hda_verb ad1986a_hp_init_verbs[] = { - {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT}, - {} -}; - - -/* models */ -enum { - AD1986A_AUTO, - AD1986A_6STACK, - AD1986A_3STACK, - AD1986A_LAPTOP, - AD1986A_LAPTOP_EAPD, - AD1986A_LAPTOP_AUTOMUTE, - AD1986A_MODELS -}; - -static const char * const ad1986a_models[AD1986A_MODELS] = { - [AD1986A_AUTO] = "auto", - [AD1986A_6STACK] = "6stack", - [AD1986A_3STACK] = "3stack", - [AD1986A_LAPTOP] = "laptop", - [AD1986A_LAPTOP_EAPD] = "laptop-eapd", - [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute", -}; - -static const struct snd_pci_quirk ad1986a_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD), - SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK), - SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK), - SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK), - SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK), - SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), - SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), - {} -}; - -#ifdef CONFIG_PM -static const struct hda_amp_list ad1986a_loopbacks[] = { - { 0x13, HDA_OUTPUT, 0 }, /* Mic */ - { 0x14, HDA_OUTPUT, 0 }, /* Phone */ - { 0x15, HDA_OUTPUT, 0 }, /* CD */ - { 0x16, HDA_OUTPUT, 0 }, /* Aux */ - { 0x17, HDA_OUTPUT, 0 }, /* Line */ - { } /* end */ -}; -#endif - -static int is_jack_available(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int conf = snd_hda_codec_get_pincfg(codec, nid); - return get_defcfg_connect(conf) != AC_JACK_PORT_NONE; -} -#endif /* ENABLE_AD_STATIC_QUIRKS */ - static int alloc_ad_spec(struct hda_codec *codec) { struct ad198x_spec *spec; @@ -1114,6 +227,9 @@ enum { AD1986A_FIXUP_INV_JACK_DETECT, AD1986A_FIXUP_ULTRA, AD1986A_FIXUP_SAMSUNG, + AD1986A_FIXUP_3STACK, + AD1986A_FIXUP_LAPTOP, + AD1986A_FIXUP_LAPTOP_IMIC, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -1139,18 +255,68 @@ static const struct hda_fixup ad1986a_fixups[] = { {} }, }, + [AD1986A_FIXUP_3STACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x02214021 }, /* headphone */ + { 0x1b, 0x01014011 }, /* front */ + { 0x1c, 0x01013012 }, /* surround */ + { 0x1d, 0x01019015 }, /* clfe */ + { 0x1e, 0x411111f0 }, /* N/A */ + { 0x1f, 0x02a190f0 }, /* mic */ + { 0x20, 0x018130f0 }, /* line-in */ + {} + }, + }, + [AD1986A_FIXUP_LAPTOP] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x02214021 }, /* headphone */ + { 0x1b, 0x90170110 }, /* speaker */ + { 0x1c, 0x411111f0 }, /* N/A */ + { 0x1d, 0x411111f0 }, /* N/A */ + { 0x1e, 0x411111f0 }, /* N/A */ + { 0x1f, 0x02a191f0 }, /* mic */ + { 0x20, 0x411111f0 }, /* N/A */ + {} + }, + }, + [AD1986A_FIXUP_LAPTOP_IMIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1d, 0x90a7013e }, /* int mic */ + {} + }, + .chained_before = 1, + .chain_id = AD1986A_FIXUP_LAPTOP, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC), + 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(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), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT), + SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK), + SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK), + {} +}; + +static const struct hda_model_fixup ad1986a_fixup_models[] = { + { .id = AD1986A_FIXUP_3STACK, .name = "3stack" }, + { .id = AD1986A_FIXUP_LAPTOP, .name = "laptop" }, + { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-imic" }, + { .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-eapd" }, /* alias */ {} }; /* */ -static int ad1986a_parse_auto_config(struct hda_codec *codec) +static int patch_ad1986a(struct hda_codec *codec) { int err; struct ad198x_spec *spec; @@ -1175,7 +341,8 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec) */ spec->gen.multiout.no_share_stream = 1; - snd_hda_pick_fixup(codec, NULL, ad1986a_fixup_tbl, ad1986a_fixups); + snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl, + ad1986a_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); err = ad198x_parse_auto_config(codec); @@ -1189,128 +356,6 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec) return 0; } -#ifdef ENABLE_AD_STATIC_QUIRKS -static int patch_ad1986a(struct hda_codec *codec) -{ - struct ad198x_spec *spec; - int err, board_config; - - board_config = snd_hda_check_board_config(codec, AD1986A_MODELS, - ad1986a_models, - ad1986a_cfg_tbl); - if (board_config < 0) { - printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n", - codec->chip_name); - board_config = AD1986A_AUTO; - } - - if (board_config == AD1986A_AUTO) - return ad1986a_parse_auto_config(codec); - - err = alloc_ad_spec(codec); - if (err < 0) - return err; - spec = codec->spec; - - err = snd_hda_attach_beep_device(codec, 0x19); - if (err < 0) { - ad198x_free(codec); - return err; - } - set_beep_amp(spec, 0x18, 0, HDA_OUTPUT); - - spec->multiout.max_channels = 6; - spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids); - spec->multiout.dac_nids = ad1986a_dac_nids; - spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT; - spec->num_adc_nids = 1; - spec->adc_nids = ad1986a_adc_nids; - spec->capsrc_nids = ad1986a_capsrc_nids; - spec->input_mux = &ad1986a_capture_source; - spec->num_mixers = 1; - spec->mixers[0] = ad1986a_mixers; - spec->num_init_verbs = 1; - spec->init_verbs[0] = ad1986a_init_verbs; -#ifdef CONFIG_PM - spec->loopback.amplist = ad1986a_loopbacks; -#endif - spec->vmaster_nid = 0x1b; - codec->inv_eapd = 1; /* AD1986A has the inverted EAPD implementation */ - - codec->patch_ops = ad198x_patch_ops; - - /* override some parameters */ - switch (board_config) { - case AD1986A_3STACK: - spec->num_mixers = 2; - spec->mixers[1] = ad1986a_3st_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1986a_ch2_init; - spec->channel_mode = ad1986a_modes; - spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); - spec->need_dac_fix = 1; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - break; - case AD1986A_LAPTOP: - spec->mixers[0] = ad1986a_laptop_mixers; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - break; - case AD1986A_LAPTOP_EAPD: - spec->num_mixers = 3; - spec->mixers[0] = ad1986a_laptop_master_mixers; - spec->mixers[1] = ad1986a_laptop_eapd_mixers; - spec->mixers[2] = ad1986a_laptop_intmic_mixers; - spec->num_init_verbs = 2; - spec->init_verbs[1] = ad1986a_eapd_init_verbs; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - if (!is_jack_available(codec, 0x25)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_laptop_eapd_capture_source; - break; - case AD1986A_LAPTOP_AUTOMUTE: - spec->num_mixers = 3; - spec->mixers[0] = ad1986a_automute_master_mixers; - spec->mixers[1] = ad1986a_laptop_eapd_mixers; - spec->mixers[2] = ad1986a_laptop_intmic_mixers; - spec->num_init_verbs = 3; - spec->init_verbs[1] = ad1986a_eapd_init_verbs; - spec->init_verbs[2] = ad1986a_hp_init_verbs; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = 1; - spec->multiout.dac_nids = ad1986a_laptop_dac_nids; - if (!is_jack_available(codec, 0x25)) - spec->multiout.dig_out_nid = 0; - spec->input_mux = &ad1986a_laptop_eapd_capture_source; - codec->patch_ops.unsol_event = ad1986a_hp_unsol_event; - codec->patch_ops.init = ad1986a_hp_init; - /* Lenovo N100 seems to report the reversed bit - * for HP jack-sensing - */ - spec->inv_jack_detect = 1; - break; - } - - /* AD1986A has a hardware problem that it can't share a stream - * with multiple output pins. The copy of front to surrounds - * causes noisy or silent outputs at a certain timing, e.g. - * changing the volume. - * So, let's disable the shared stream. - */ - spec->multiout.no_share_stream = 1; - - codec->no_trigger_sense = 1; - codec->no_sticky_stream = 1; - - return 0; -} -#else /* ENABLE_AD_STATIC_QUIRKS */ -#define patch_ad1986a ad1986a_parse_auto_config -#endif /* ENABLE_AD_STATIC_QUIRKS */ /* * AD1983 specific -- cgit v1.2.3 From 384f778fd924cc843acf93c23f52cb168cb3f02a Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:27:53 +0200 Subject: ALSA: hdspm - Add missing defines for RME AIO and RayDAT The driver did not support all possible configurations. These defines will be used by later commits to add the missing functionality. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index bd501931ee23..a0fc961bc347 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -258,6 +258,25 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wclk_sel (1<<30) +/* additional control register bits for AIO*/ +#define HDSPM_c0_Wck48 0x20 /* also RayDAT */ +#define HDSPM_c0_Input0 0x1000 +#define HDSPM_c0_Input1 0x2000 +#define HDSPM_c0_Spdif_Opt 0x4000 +#define HDSPM_c0_Pro 0x8000 +#define HDSPM_c0_clr_tms 0x10000 +#define HDSPM_c0_AEB1 0x20000 +#define HDSPM_c0_AEB2 0x40000 +#define HDSPM_c0_LineOut 0x80000 +#define HDSPM_c0_AD_GAIN0 0x100000 +#define HDSPM_c0_AD_GAIN1 0x200000 +#define HDSPM_c0_DA_GAIN0 0x400000 +#define HDSPM_c0_DA_GAIN1 0x800000 +#define HDSPM_c0_PH_GAIN0 0x1000000 +#define HDSPM_c0_PH_GAIN1 0x2000000 +#define HDSPM_c0_Sym6db 0x4000000 + + /* --- bit helper defines */ #define HDSPM_LatencyMask (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2) #define HDSPM_FrequencyMask (HDSPM_Frequency0|HDSPM_Frequency1|\ -- cgit v1.2.3 From b2ed6326874b1bf5410871d83df4086a395ab13b Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:27:54 +0200 Subject: ALSA: hdspm - Introduce hdspm_is_raydat_or_aio() RME RayDAT and AIO cards are new designs with different register settings. Since we need to distinguish them from older cards multiple times in the driver, refactor the code into a separate helper function. No functional change intended. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a0fc961bc347..32a87dcecfa3 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1011,6 +1011,12 @@ static inline int HDSPM_bit2freq(int n) return bit2freq_tab[n]; } +static bool hdspm_is_raydat_or_aio(struct hdspm *hdspm) +{ + return ((AIO == hdspm->io_type) || (RayDAT == hdspm->io_type)); +} + + /* Write/read to/from HDSPM with Adresses in Bytes not words but only 32Bit writes are allowed */ @@ -5142,9 +5148,8 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm) all_in_all_mixer(hdspm, 0 * UNITY_GAIN); - if (hdspm->io_type == AIO || hdspm->io_type == RayDAT) { + if (hdspm_is_raydat_or_aio(hdspm)) hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register); - } /* set a default rate so that the channel map is set up. */ hdspm_set_rate(hdspm, 48000, 1); -- cgit v1.2.3 From ce13f3f33a32895da9304a9f9cb865f337dd0933 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:27:55 +0200 Subject: ALSA: hdspm - Augment HDSPM_TOGGLE_SETTING for AIO/RayDAT The HDSPM_TOGGLE_SETTING functions alter the control_register on older cards. On newer cards (AIO/RayDAT), they have to operate on the settings_register instead. This patch augments the existing functions to work with AIO/RayDAT, too. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 27 +++++++++++++++++++++++---- 1 file changed, 23 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 32a87dcecfa3..118d727150e7 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3092,16 +3092,35 @@ static int snd_hdspm_get_tco_ltc_frames(struct snd_kcontrol *kcontrol, static int hdspm_toggle_setting(struct hdspm *hdspm, u32 regmask) { - return (hdspm->control_register & regmask) ? 1 : 0; + u32 reg; + + if (hdspm_is_raydat_or_aio(hdspm)) + reg = hdspm->settings_register; + else + reg = hdspm->control_register; + + return (reg & regmask) ? 1 : 0; } static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out) { + u32 *reg; + u32 target_reg; + + if (hdspm_is_raydat_or_aio(hdspm)) { + reg = &(hdspm->settings_register); + target_reg = HDSPM_WR_SETTINGS; + } else { + reg = &(hdspm->control_register); + target_reg = HDSPM_controlRegister; + } + if (out) - hdspm->control_register |= regmask; + *reg |= regmask; else - hdspm->control_register &= ~regmask; - hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register); + *reg &= ~regmask; + + hdspm_write(hdspm, target_reg, *reg); return 0; } -- cgit v1.2.3 From 34be7ebbb4488818a2c413290b7b5835173fe44d Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:27:56 +0200 Subject: ALSA: hdspm - Drop duplicate code in hdspm_set_system_clock_mode() hdspm_set_system_clock_mode() is almost a one-by-one copy of hdspm_set_toggle_setting(). To improve code quality, remove the duplication. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 25 +++++-------------------- 1 file changed, 5 insertions(+), 20 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 118d727150e7..631c54601013 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -995,6 +995,7 @@ static inline void snd_hdspm_initialize_midi_flush(struct hdspm *hdspm); static inline int hdspm_get_pll_freq(struct hdspm *hdspm); static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm); static int hdspm_autosync_ref(struct hdspm *hdspm); +static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out); static int snd_hdspm_set_defaults(struct hdspm *hdspm); static int hdspm_system_clock_mode(struct hdspm *hdspm); static void hdspm_set_sgbuf(struct hdspm *hdspm, @@ -2384,26 +2385,10 @@ static int hdspm_system_clock_mode(struct hdspm *hdspm) **/ static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode) { - switch (hdspm->io_type) { - case AIO: - case RayDAT: - if (0 == mode) - hdspm->settings_register |= HDSPM_c0Master; - else - hdspm->settings_register &= ~HDSPM_c0Master; - - hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register); - break; - - default: - if (0 == mode) - hdspm->control_register |= HDSPM_ClockModeMaster; - else - hdspm->control_register &= ~HDSPM_ClockModeMaster; - - hdspm_write(hdspm, HDSPM_controlRegister, - hdspm->control_register); - } + hdspm_set_toggle_setting(hdspm, + (hdspm_is_raydat_or_aio(hdspm)) ? + HDSPM_c0Master : HDSPM_ClockModeMaster, + (0 == mode)); } -- cgit v1.2.3 From 11a5cd3c950ee27b165b5c170e588dff22cadeca Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:27:57 +0200 Subject: ALSA: hdspm - Add S/PDIF and WCK48 controls for RME RayDAT This commit adds new ALSA controls to send single-speed WordClock and S/PDIF-Professional on RME RayDAT cards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 631c54601013..4a3a82249153 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4364,7 +4364,9 @@ static struct snd_kcontrol_new snd_hdspm_controls_raydat[] = { HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT3 Frequency", 5), HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT4 Frequency", 6), HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 7), - HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8) + HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8), + HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro), + HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48) }; static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = { -- cgit v1.2.3 From fb0f121e0f346bec45810a9439e936ae62fd2441 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:27:58 +0200 Subject: ALSA: hdspm - Add S/PDIF, XLR, WCK48 and ADAT-in controls for RME AIO cards This commit adds the following ALSA controls: - S/PDIF Out Optical to switch S/PDIF Out from coaxial to optical - S/PDIF Out Professional to send the Pro bit in the output stream - ADAT-Internal to enable ADAT/TDIF Expansion Board (AEB/TEB) - XLR Breakout Cable if analogue I/O uses the XLR breakout cable - WCK48 to force WordClock to the 32-48kHz range (single speed) if the card is operating at higher frequencies Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 4a3a82249153..15f1e7b37c9f 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4327,7 +4327,12 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = { HDSPM_AUTOSYNC_SAMPLE_RATE("SPDIF Frequency", 2), HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT Frequency", 3), HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 4), - HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5) + HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5), + HDSPM_TOGGLE_SETTING("S/PDIF Out Optical", HDSPM_c0_Spdif_Opt), + HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro), + HDSPM_TOGGLE_SETTING("ADAT internal (AEB/TEB)", HDSPM_c0_AEB1), + HDSPM_TOGGLE_SETTING("XLR Breakout Cable", HDSPM_c0_Sym6db), + HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48) /* HDSPM_INPUT_SELECT("Input Select", 0), -- cgit v1.2.3 From 8cea57104273909ab0825df48149840aad9d2b14 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:27:59 +0200 Subject: ALSA: hdspm - Refactor ENUMERATED_CTL_INFO into function ENUMERATED_CTL_INFO is a macro, so the binary code is generated multiple times. To avoid code duplication, refactor the involved functionality into a function and make ENUMERATED_CTL_INFO a call to this function. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 22 ++++++++++++++-------- 1 file changed, 14 insertions(+), 8 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 15f1e7b37c9f..b271853e2e52 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2221,16 +2221,22 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx) return (status >> (idx*4)) & 0xF; } -#define ENUMERATED_CTL_INFO(info, texts) \ -{ \ - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; \ - uinfo->count = 1; \ - uinfo->value.enumerated.items = ARRAY_SIZE(texts); \ - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) \ - uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; \ - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); \ +static void snd_hdspm_set_infotext(struct snd_ctl_elem_info *uinfo, + char **texts, const int count) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = count; + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = + uinfo->value.enumerated.items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); } +#define ENUMERATED_CTL_INFO(info, texts) \ + snd_hdspm_set_infotext(info, texts, ARRAY_SIZE(texts)) + #define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \ -- cgit v1.2.3 From acf14767e17ab7ee8b6213f9e56d07d9ffa033da Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:00 +0200 Subject: ALSA: hdspm - Introduce generic AIO tristate control AIO cards offer at least four individual settings options with three states each. Those settings are represented as two bits in the settings register with the following meaning: 0*some_base_bit --> Option value 0 1*some_base_bit --> Option value 1 2*some_base_bit --> Option value 2 3*some_base_bit --> mask to select the two involved bits This patch adds a generic ALSA control macro for such a value-to-bit pattern mapping. It will be used in a later commit to expose four new controls. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 78 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 78 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b271853e2e52..d9532c42e2ce 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3348,6 +3348,84 @@ static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol, return change; } +#define HDSPM_CONTROL_TRISTATE(xname, xindex) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .private_value = xindex, \ + .info = snd_hdspm_info_tristate, \ + .get = snd_hdspm_get_tristate, \ + .put = snd_hdspm_put_tristate \ +} + +static int hdspm_tristate(struct hdspm *hdspm, u32 regmask) +{ + u32 reg = hdspm->settings_register & (regmask * 3); + return reg / regmask; +} + +static int hdspm_set_tristate(struct hdspm *hdspm, int mode, u32 regmask) +{ + hdspm->settings_register &= ~(regmask * 3); + hdspm->settings_register |= (regmask * mode); + hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register); + + return 0; +} + +static int snd_hdspm_info_tristate(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + u32 regmask = kcontrol->private_value; + + static char *texts_spdif[] = { "Optical", "Coaxial", "Internal" }; + static char *texts_levels[] = { "Hi Gain", "+4 dBu", "-10 dBV" }; + + switch (regmask) { + case HDSPM_c0_Input0: + ENUMERATED_CTL_INFO(uinfo, texts_spdif); + break; + default: + ENUMERATED_CTL_INFO(uinfo, texts_levels); + break; + } + return 0; +} + +static int snd_hdspm_get_tristate(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); + u32 regmask = kcontrol->private_value; + + spin_lock_irq(&hdspm->lock); + ucontrol->value.enumerated.item[0] = hdspm_tristate(hdspm, regmask); + spin_unlock_irq(&hdspm->lock); + return 0; +} + +static int snd_hdspm_put_tristate(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); + u32 regmask = kcontrol->private_value; + int change; + int val; + + if (!snd_hdspm_use_is_exclusive(hdspm)) + return -EBUSY; + val = ucontrol->value.integer.value[0]; + if (val < 0) + val = 0; + if (val > 2) + val = 2; + + spin_lock_irq(&hdspm->lock); + change = val != hdspm_tristate(hdspm, regmask); + hdspm_set_tristate(hdspm, val, regmask); + spin_unlock_irq(&hdspm->lock); + return change; +} + #define HDSPM_MADI_SPEEDMODE(xname, xindex) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ -- cgit v1.2.3 From 42f4c12dcf46cbca8b7bb17610c0cb7ffbd7ab2e Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:01 +0200 Subject: ALSA: hdspm - Enable AD/DA/PH gains and S/PDIF-Input select on AIO This patch uses the newly introduced HDSPM_CONTROL_TRISTATE functions to create and expose the following ALSA controls: - Gain selection for Input, Output and Phones (HiGain, +4dBu, -10dbV) - S/PDIF Input select (Coaxial, Optical, Internal) Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d9532c42e2ce..778fc23105fa 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4412,11 +4412,15 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = { HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT Frequency", 3), HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 4), HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5), + HDSPM_CONTROL_TRISTATE("S/PDIF Input", HDSPM_c0_Input0), HDSPM_TOGGLE_SETTING("S/PDIF Out Optical", HDSPM_c0_Spdif_Opt), HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro), HDSPM_TOGGLE_SETTING("ADAT internal (AEB/TEB)", HDSPM_c0_AEB1), HDSPM_TOGGLE_SETTING("XLR Breakout Cable", HDSPM_c0_Sym6db), - HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48) + HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48), + HDSPM_CONTROL_TRISTATE("Input Level", HDSPM_c0_AD_GAIN0), + HDSPM_CONTROL_TRISTATE("Output Level", HDSPM_c0_DA_GAIN0), + HDSPM_CONTROL_TRISTATE("Phones Level", HDSPM_c0_PH_GAIN0) /* HDSPM_INPUT_SELECT("Input Select", 0), -- cgit v1.2.3 From 3de9db264cef4bc984f928e08cccf36304f30d0a Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:02 +0200 Subject: ALSA: hdspm - Add support for AEBs on RME AIO AIO cards allow to use AEB (Analogue Expansion Boards) to add four input and/or output channels. This patch adds the necessary code to detect and enable the additional I/O channels. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 54 ++++++++++++++++++++++++++++++++--------------- 1 file changed, 37 insertions(+), 17 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 778fc23105fa..ad416365dc68 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -648,7 +648,8 @@ static char *texts_ports_aio_in_ss[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", - "ADAT.7", "ADAT.8" + "ADAT.7", "ADAT.8", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ss[] = { @@ -657,14 +658,16 @@ static char *texts_ports_aio_out_ss[] = { "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6", "ADAT.7", "ADAT.8", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_ds[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_ds[] = { @@ -672,14 +675,16 @@ static char *texts_ports_aio_out_ds[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_in_qs[] = { "Analogue.L", "Analogue.R", "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", - "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4" + "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aio_out_qs[] = { @@ -687,7 +692,8 @@ static char *texts_ports_aio_out_qs[] = { "AES.L", "AES.R", "SPDIF.L", "SPDIF.R", "ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", - "Phone.L", "Phone.R" + "Phone.L", "Phone.R", + "AEB.1", "AEB.2", "AEB.3", "AEB.4" }; static char *texts_ports_aes32[] = { @@ -764,8 +770,8 @@ static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in, */ 10, 11, /* spdif in */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT in */ - -1, -1, - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -779,7 +785,8 @@ static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 13, 14, 15, 16, 17, 18, 19, /* ADAT out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -792,7 +799,8 @@ static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 14, 16, 18, /* adat in */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -807,7 +815,7 @@ static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 14, 16, 18, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -821,7 +829,8 @@ static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = { 8, 9, /* aes in */ 10, 11, /* spdif in */ 12, 16, /* adat in */ - -1, -1, -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -836,7 +845,8 @@ static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = { 10, 11, /* spdif out */ 12, 16, /* adat out */ 6, 7, /* phone out */ - -1, -1, -1, -1, -1, -1, + 2, 3, 4, 5, /* AEB */ + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, @@ -6602,10 +6612,6 @@ static int snd_hdspm_create(struct snd_card *card, break; case AIO: - if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { - snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n"); - } - hdspm->ss_in_channels = AIO_IN_SS_CHANNELS; hdspm->ds_in_channels = AIO_IN_DS_CHANNELS; hdspm->qs_in_channels = AIO_IN_QS_CHANNELS; @@ -6613,6 +6619,20 @@ static int snd_hdspm_create(struct snd_card *card, hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS; hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS; + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) { + snd_printk(KERN_INFO "HDSPM: AEB input board found\n"); + hdspm->ss_in_channels += 4; + hdspm->ds_in_channels += 4; + hdspm->qs_in_channels += 4; + } + + if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) { + snd_printk(KERN_INFO "HDSPM: AEB output board found\n"); + hdspm->ss_out_channels += 4; + hdspm->ds_out_channels += 4; + hdspm->qs_out_channels += 4; + } + hdspm->channel_map_out_ss = channel_map_aio_out_ss; hdspm->channel_map_out_ds = channel_map_aio_out_ds; hdspm->channel_map_out_qs = channel_map_aio_out_qs; -- cgit v1.2.3 From 1cb7dbf489f9985b7a117e34d00f20799adb138a Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:03 +0200 Subject: ALSA: hdspm - Fix S/PDIF Sync status and frequency on RME AIO This is a left-over mistake from old code, the correct register offset is provided in kcontrol->private_value, not in the index. Cf. RayDAT case, where it has already been corrected. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ad416365dc68..06e69de0801e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2312,7 +2312,7 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, default: ucontrol->value.enumerated.item[0] = hdspm_get_s1_sample_rate(hdspm, - ucontrol->id.index-1); + kcontrol->private_value-1); } break; @@ -3930,7 +3930,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol, case 5: /* SYNC IN */ val = hdspm_sync_in_sync_check(hdspm); break; default: - val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1); + val = hdspm_s1_sync_check(hdspm, + kcontrol->private_value-1); } break; -- cgit v1.2.3 From 5760107c8263cf518968ece65453b7d9b8ca3d0a Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:04 +0200 Subject: ALSA: hdspm - Create TCO readout function This patch separates the TCO bits from snd_hdspm_proc_read_madi(), so the new function can later be shared between MADI and AES32 cards. It's essentially only moving code around, no new functionality. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 137 ++++++++++++++++++++++++++-------------------- 1 file changed, 77 insertions(+), 60 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 06e69de0801e..58b21048c2e8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4636,77 +4636,22 @@ static int snd_hdspm_create_controls(struct snd_card *card, ------------------------------------------------------------*/ static void -snd_hdspm_proc_read_madi(struct snd_info_entry * entry, - struct snd_info_buffer *buffer) +snd_hdspm_proc_read_tco(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { struct hdspm *hdspm = entry->private_data; - unsigned int status, status2, control, freq; - - char *pref_sync_ref; - char *autosync_ref; - char *system_clock_mode; - char *insel; - int x, x2; - - /* TCO stuff */ + unsigned int status, control; int a, ltc, frames, seconds, minutes, hours; unsigned int period; u64 freq_const = 0; u32 rate; + snd_iprintf(buffer, "--- TCO ---\n"); + status = hdspm_read(hdspm, HDSPM_statusRegister); - status2 = hdspm_read(hdspm, HDSPM_statusRegister2); control = hdspm->control_register; - freq = hdspm_read(hdspm, HDSPM_timecodeRegister); - - snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n", - hdspm->card_name, hdspm->card->number + 1, - hdspm->firmware_rev, - (status2 & HDSPM_version0) | - (status2 & HDSPM_version1) | (status2 & - HDSPM_version2)); - - snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n", - (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF, - hdspm->serial); - - snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n", - hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase); - - snd_iprintf(buffer, "--- System ---\n"); - snd_iprintf(buffer, - "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n", - status & HDSPM_audioIRQPending, - (status & HDSPM_midi0IRQPending) ? 1 : 0, - (status & HDSPM_midi1IRQPending) ? 1 : 0, - hdspm->irq_count); - snd_iprintf(buffer, - "HW pointer: id = %d, rawptr = %d (%d->%d) " - "estimated= %ld (bytes)\n", - ((status & HDSPM_BufferID) ? 1 : 0), - (status & HDSPM_BufferPositionMask), - (status & HDSPM_BufferPositionMask) % - (2 * (int)hdspm->period_bytes), - ((status & HDSPM_BufferPositionMask) - 64) % - (2 * (int)hdspm->period_bytes), - (long) hdspm_hw_pointer(hdspm) * 4); - snd_iprintf(buffer, - "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n", - hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF, - hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF, - hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF, - hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF); - snd_iprintf(buffer, - "MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n", - hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF, - hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF); - snd_iprintf(buffer, - "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, " - "status2=0x%x\n", - hdspm->control_register, hdspm->control2_register, - status, status2); if (status & HDSPM_tco_detect) { snd_iprintf(buffer, "TCO module detected.\n"); a = hdspm_read(hdspm, HDSPM_RD_TCO+4); @@ -4800,6 +4745,75 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, } else { snd_iprintf(buffer, "No TCO module detected.\n"); } +} + +static void +snd_hdspm_proc_read_madi(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct hdspm *hdspm = entry->private_data; + unsigned int status, status2, control, freq; + + char *pref_sync_ref; + char *autosync_ref; + char *system_clock_mode; + char *insel; + int x, x2; + + status = hdspm_read(hdspm, HDSPM_statusRegister); + status2 = hdspm_read(hdspm, HDSPM_statusRegister2); + control = hdspm->control_register; + freq = hdspm_read(hdspm, HDSPM_timecodeRegister); + + snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n", + hdspm->card_name, hdspm->card->number + 1, + hdspm->firmware_rev, + (status2 & HDSPM_version0) | + (status2 & HDSPM_version1) | (status2 & + HDSPM_version2)); + + snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n", + (hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF, + hdspm->serial); + + snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n", + hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase); + + snd_iprintf(buffer, "--- System ---\n"); + + snd_iprintf(buffer, + "IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n", + status & HDSPM_audioIRQPending, + (status & HDSPM_midi0IRQPending) ? 1 : 0, + (status & HDSPM_midi1IRQPending) ? 1 : 0, + hdspm->irq_count); + snd_iprintf(buffer, + "HW pointer: id = %d, rawptr = %d (%d->%d) " + "estimated= %ld (bytes)\n", + ((status & HDSPM_BufferID) ? 1 : 0), + (status & HDSPM_BufferPositionMask), + (status & HDSPM_BufferPositionMask) % + (2 * (int)hdspm->period_bytes), + ((status & HDSPM_BufferPositionMask) - 64) % + (2 * (int)hdspm->period_bytes), + (long) hdspm_hw_pointer(hdspm) * 4); + + snd_iprintf(buffer, + "MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n", + hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF, + hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF, + hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF, + hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF); + snd_iprintf(buffer, + "MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n", + hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF, + hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF); + snd_iprintf(buffer, + "Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, " + "status2=0x%x\n", + hdspm->control_register, hdspm->control2_register, + status, status2); + snd_iprintf(buffer, "--- Settings ---\n"); @@ -4903,6 +4917,9 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry, (status & HDSPM_RX_64ch) ? "64 channels" : "56 channels"); + /* call readout function for TCO specific status */ + snd_hdspm_proc_read_tco(entry, buffer); + snd_iprintf(buffer, "\n"); } -- cgit v1.2.3 From b0bf550476a5a6238baf1309ba913ca9f7a379ba Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:05 +0200 Subject: ALSA: hdspm - AES32: Fix TCO sync check reporting HDSPM_tco_lock and HDSPM_tcoLock were too close, so the previous code didn't honour the difference between the two. Let's be more verbose and use HDSPM_tcoLockMadi for MADI cards, HDSPM_tcoLockAes for AES(32) and fix the code that makes use of both. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 25 ++++++++++++++++++------- 1 file changed, 18 insertions(+), 7 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 58b21048c2e8..bdd8c7742a1f 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -360,11 +360,11 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_madiLock (1<<3) /* MADI Locked =1, no=0 */ #define HDSPM_madiSync (1<<18) /* MADI is in sync */ -#define HDSPM_tcoLock 0x00000020 /* Optional TCO locked status FOR HDSPe MADI! */ -#define HDSPM_tcoSync 0x10000000 /* Optional TCO sync status */ +#define HDSPM_tcoLockMadi 0x00000020 /* Optional TCO locked status for HDSPe MADI*/ +#define HDSPM_tcoSync 0x10000000 /* Optional TCO sync status for HDSPe MADI and AES32!*/ -#define HDSPM_syncInLock 0x00010000 /* Sync In lock status FOR HDSPe MADI! */ -#define HDSPM_syncInSync 0x00020000 /* Sync In sync status FOR HDSPe MADI! */ +#define HDSPM_syncInLock 0x00010000 /* Sync In lock status for HDSPe MADI! */ +#define HDSPM_syncInSync 0x00020000 /* Sync In sync status for HDSPe MADI! */ #define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */ /* since 64byte accurate, last 6 bits are not used */ @@ -382,7 +382,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); * Interrupt */ #define HDSPM_tco_detect 0x08000000 -#define HDSPM_tco_lock 0x20000000 +#define HDSPM_tcoLockAes 0x20000000 /* Optional TCO locked status for HDSPe AES */ #define HDSPM_s2_tco_detect 0x00000040 #define HDSPM_s2_AEBO_D 0x00000080 @@ -480,7 +480,9 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_AES32_AUTOSYNC_FROM_AES6 6 #define HDSPM_AES32_AUTOSYNC_FROM_AES7 7 #define HDSPM_AES32_AUTOSYNC_FROM_AES8 8 -#define HDSPM_AES32_AUTOSYNC_FROM_NONE 9 +#define HDSPM_AES32_AUTOSYNC_FROM_TCO 9 +#define HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN 10 +#define HDSPM_AES32_AUTOSYNC_FROM_NONE 11 /* status2 */ /* HDSPM_LockAES_bit is given by HDSPM_LockAES >> (AES# - 1) */ @@ -3868,9 +3870,18 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm) if (hdspm->tco) { switch (hdspm->io_type) { case MADI: + status = hdspm_read(hdspm, HDSPM_statusRegister); + if (status & HDSPM_tcoLockMadi) { + if (status & HDSPM_tcoSync) + return 2; + else + return 1; + } + return 0; + break; case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); - if (status & HDSPM_tcoLock) { + if (status & HDSPM_tcoLockAes) { if (status & HDSPM_tcoSync) return 2; else -- cgit v1.2.3 From e71b95ad71e3ee44ec634e242b186e3ff03bd459 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:06 +0200 Subject: ALSA: hdspm - Cosmetics, no real change This patch does nothing, it's sole intent is to clean up the code. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index bdd8c7742a1f..d95100e7bd22 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2926,7 +2926,7 @@ static int hdspm_autosync_ref(struct hdspm *hdspm) case HDSPM_SelSyncRef_NVALID: return HDSPM_AUTOSYNC_FROM_NONE; default: - return 0; + return HDSPM_AUTOSYNC_FROM_NONE; } } @@ -5260,7 +5260,7 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm) case AES32: hdspm->control_register = - HDSPM_ClockModeMaster | /* Master Cloack Mode on */ + HDSPM_ClockModeMaster | /* Master Clock Mode on */ hdspm_encode_latency(7) | /* latency max=8192samples */ HDSPM_SyncRef0 | /* AES1 is syncclock */ HDSPM_LineOut | /* Analog output in */ @@ -6737,7 +6737,7 @@ static int snd_hdspm_create(struct snd_card *card, if (NULL != hdspm->tco) { hdspm_tco_write(hdspm); } - snd_printk(KERN_INFO "HDSPM: MADI TCO module found\n"); + snd_printk(KERN_INFO "HDSPM: MADI/AES TCO module found\n"); } else { hdspm->tco = NULL; } @@ -6752,10 +6752,12 @@ static int snd_hdspm_create(struct snd_card *card, case AES32: if (hdspm->tco) { hdspm->texts_autosync = texts_autosync_aes_tco; - hdspm->texts_autosync_items = 10; + hdspm->texts_autosync_items = + ARRAY_SIZE(texts_autosync_aes_tco); } else { hdspm->texts_autosync = texts_autosync_aes; - hdspm->texts_autosync_items = 9; + hdspm->texts_autosync_items = + ARRAY_SIZE(texts_autosync_aes); } break; -- cgit v1.2.3 From 3c32de58ae9a3d534ba1a66274bf43631e36eb5c Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:07 +0200 Subject: ALSA: hdspm - AIO: Drop superfluous HDSPM_AUTOSYNC_REF The HDSPM_AUTOSYNC_REF macro is only implemented for MADI and AES32 cards, so it doesn't make sense to call it on AIO boards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d95100e7bd22..d1e05828cee6 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4419,7 +4419,6 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = { HDSPM_INTERNAL_CLOCK("Internal Clock", 0), HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0), HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0), - HDSPM_AUTOSYNC_REF("AutoSync Reference", 0), HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0), HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0), HDSPM_SYNC_CHECK("WC SyncCheck", 0), -- cgit v1.2.3 From db2d1a913d838ecfab5b903508bcdd4e4ad42419 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:08 +0200 Subject: ALSA: hdspm - AES32: Add TCO and Sync-In text entries Provide the text for the two new clock options "TCO" and "Sync In" on AES32 cards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d1e05828cee6..8e6ce1473333 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -561,10 +561,13 @@ static char *hdspm_speed_names[] = { "single", "double", "quad" }; static char *texts_autosync_aes_tco[] = { "Word Clock", "AES1", "AES2", "AES3", "AES4", "AES5", "AES6", "AES7", "AES8", - "TCO" }; + "TCO", "Sync In" +}; static char *texts_autosync_aes[] = { "Word Clock", "AES1", "AES2", "AES3", "AES4", - "AES5", "AES6", "AES7", "AES8" }; + "AES5", "AES6", "AES7", "AES8", + "Sync In" +}; static char *texts_autosync_madi_tco[] = { "Word Clock", "MADI", "TCO", "Sync In" }; static char *texts_autosync_madi[] = { "Word Clock", @@ -2941,11 +2944,11 @@ static int snd_hdspm_info_autosync_ref(struct snd_kcontrol *kcontrol, if (AES32 == hdspm->io_type) { static char *texts[] = { "WordClock", "AES1", "AES2", "AES3", - "AES4", "AES5", "AES6", "AES7", "AES8", "None"}; + "AES4", "AES5", "AES6", "AES7", "AES8", "TCO", "Sync In", "None"}; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - uinfo->value.enumerated.items = 10; + uinfo->value.enumerated.items = ARRAY_SIZE(texts); if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = -- cgit v1.2.3 From d3c36ed8e578185b752dac4277819965fa5f6879 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:09 +0200 Subject: ALSA: hdspm - Introduce hdspm_get_aes_sample_rate() Helper function to return the AES sample rate class. This class needs to be translated via HDSPM_bit2freq() to get the more common representation. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8e6ce1473333..b7702b225c3e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2224,6 +2224,23 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm) return 0; } +/** + * Returns the AES sample rate class for the given card. + **/ +static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index) +{ + int timecode; + + switch (hdspm->io_type) { + case AES32: + timecode = hdspm_read(hdspm, HDSPM_timecodeRegister); + return (timecode >> (4*index)) & 0xF; + break; + default: + break; + } + return 0; +} /** * Returns the sample rate class for input source for -- cgit v1.2.3 From 5b266354b91087d8f1b1d1b6853a2c012f3e1518 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:10 +0200 Subject: ALSA: hdspm - Add prototype declarations This patch only introduces prototype declarations, no real change. The functions themselves are already present. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b7702b225c3e..367dd411247d 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1017,6 +1017,17 @@ static void hdspm_set_sgbuf(struct hdspm *hdspm, struct snd_pcm_substream *substream, unsigned int reg, int channels); +static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx); +static int hdspm_wc_sync_check(struct hdspm *hdspm); +static int hdspm_tco_sync_check(struct hdspm *hdspm); +static int hdspm_sync_in_sync_check(struct hdspm *hdspm); + +static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index); +static int hdspm_get_tco_sample_rate(struct hdspm *hdspm); +static int hdspm_get_wc_sample_rate(struct hdspm *hdspm); + + + static inline int HDSPM_bit2freq(int n) { static const int bit2freq_tab[] = { @@ -1152,10 +1163,7 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate) return rate; } -static int hdspm_tco_sync_check(struct hdspm *hdspm); -static int hdspm_sync_in_sync_check(struct hdspm *hdspm); - -/* check for external sample rate */ +/* check for external sample rate, returns the sample rate in Hz*/ static int hdspm_external_sample_rate(struct hdspm *hdspm) { unsigned int status, status2, timecode; -- cgit v1.2.3 From a57fea8ed44a2d32f8cbdd5455262aca88e72aa6 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:11 +0200 Subject: ALSA: hdspm - Enable AES32 in hdspm_get_wc_sample_rate This patch adds AES32 specific code to hdspm_get_wc_sample_rate() to query the wordclock frequency. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 367dd411247d..a69957cba0f4 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2178,6 +2178,9 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm) status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); return (status >> 16) & 0xF; break; + case AES32: + status = hdspm_read(hdspm, HDSPM_statusRegister); + return (status >> HDSPM_AES32_wcFreq_bit) & 0xF; default: break; } -- cgit v1.2.3 From 051c44fec7e250a93d8f3b6704a3ce880a11bb0f Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:12 +0200 Subject: ALSA: hdspm - Enable AES32 in hdspm_get_tco_sample_rate This patch adds AES32 specific code to hdspm_get_tco_sample_rate to query the TCO sample rate. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a69957cba0f4..c0143cf83cf3 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2204,6 +2204,9 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm) status = hdspm_read(hdspm, HDSPM_RD_STATUS_1); return (status >> 20) & 0xF; break; + case AES32: + status = hdspm_read(hdspm, HDSPM_statusRegister); + return (status >> 1) & 0xF; default: break; } -- cgit v1.2.3 From 3ac9b0acc34fbe56e2d31b8f2f7e59d45c53cb3b Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:13 +0200 Subject: ALSA: hdspm - AES32: Ignore float/int format bit As mentioned in the comment, the AES32 cards must not set the format bit, since it is used to indicate the preferred sync setting instead. We hence simply skip the corresponding part in the hw_params function. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index c0143cf83cf3..a9f4c7c4b6f4 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -5566,6 +5566,16 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream, */ + /* For AES cards, the float format bit is the same as the + * preferred sync reference. Since we don't want to break + * sync settings, we have to skip the remaining part of this + * function. + */ + if (hdspm->io_type == AES32) { + return 0; + } + + /* Switch to native float format if requested */ if (SNDRV_PCM_FORMAT_FLOAT_LE == params_format(params)) { if (!(hdspm->control_register & HDSPe_FLOAT_FORMAT)) -- cgit v1.2.3 From dbae4a0c8d8794df1a6bd7e644ed94b915f46f7e Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:14 +0200 Subject: ALSA: hdspm - AES32: Enable TCO input in hdspm_external_sample_rate() This patch adds support to read the TCO sample rate in hdspm_external_sample_rate() on RME AES(32) cards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 37 ++++++++++++++++++++++++++++--------- 1 file changed, 28 insertions(+), 9 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a9f4c7c4b6f4..80b2247dc8f9 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1176,17 +1176,36 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) timecode = hdspm_read(hdspm, HDSPM_timecodeRegister); syncref = hdspm_autosync_ref(hdspm); + switch (syncref) { + case HDSPM_AES32_AUTOSYNC_FROM_WORD: + /* Check WC sync and get sample rate */ + if (hdspm_wc_sync_check(hdspm)) + return HDSPM_bit2freq(hdspm_get_wc_sample_rate(hdspm)); + break; + + case HDSPM_AES32_AUTOSYNC_FROM_AES1: + case HDSPM_AES32_AUTOSYNC_FROM_AES2: + case HDSPM_AES32_AUTOSYNC_FROM_AES3: + case HDSPM_AES32_AUTOSYNC_FROM_AES4: + case HDSPM_AES32_AUTOSYNC_FROM_AES5: + case HDSPM_AES32_AUTOSYNC_FROM_AES6: + case HDSPM_AES32_AUTOSYNC_FROM_AES7: + case HDSPM_AES32_AUTOSYNC_FROM_AES8: + /* Check AES sync and get sample rate */ + if (hdspm_aes_sync_check(hdspm, syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)) + return HDSPM_bit2freq(hdspm_get_aes_sample_rate(hdspm, + syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)); + break; - if (syncref == HDSPM_AES32_AUTOSYNC_FROM_WORD && - status & HDSPM_AES32_wcLock) - return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF); - if (syncref >= HDSPM_AES32_AUTOSYNC_FROM_AES1 && - syncref <= HDSPM_AES32_AUTOSYNC_FROM_AES8 && - status2 & (HDSPM_LockAES >> - (syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1))) - return HDSPM_bit2freq((timecode >> (4*(syncref-HDSPM_AES32_AUTOSYNC_FROM_AES1))) & 0xF); - return 0; + case HDSPM_AES32_AUTOSYNC_FROM_TCO: + /* Check TCO sync and get sample rate */ + if (hdspm_tco_sync_check(hdspm)) + return HDSPM_bit2freq(hdspm_get_tco_sample_rate(hdspm)); + break; + default: + return 0; + } /* end switch(syncref) */ break; case MADIface: -- cgit v1.2.3 From 2d60fc7f7d3d79e5646646bb34811961f19d111a Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:15 +0200 Subject: ALSA: hdspm - AES32: Enable TCO/Sync-In in snd_hdspm_put_sync_ref() This patch enables the user to select "TCO" and "Sync In" as a preferred sync reference on RME AES(32) cards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 80b2247dc8f9..73d96269c9b2 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2954,19 +2954,20 @@ static int snd_hdspm_put_pref_sync_ref(struct snd_kcontrol *kcontrol, static int hdspm_autosync_ref(struct hdspm *hdspm) { + /* This looks at the autosync selected sync reference */ if (AES32 == hdspm->io_type) { + unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister); - unsigned int syncref = - (status >> HDSPM_AES32_syncref_bit) & 0xF; - if (syncref == 0) - return HDSPM_AES32_AUTOSYNC_FROM_WORD; - if (syncref <= 8) + unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF; + if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) && + (syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN)) { return syncref; + } return HDSPM_AES32_AUTOSYNC_FROM_NONE; + } else if (MADI == hdspm->io_type) { - /* This looks at the autosync selected sync reference */ - unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2); + unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2); switch (status2 & HDSPM_SelSyncRefMask) { case HDSPM_SelSyncRef_WORD: return HDSPM_AUTOSYNC_FROM_WORD; -- cgit v1.2.3 From 194062daba00688dfd47caaf01f3131700cd726f Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:16 +0200 Subject: ALSA: hdspm - AES32: Include TCO and Sync-In in proc output Also report TCO status and Sync-In via /proc/ on AES(32) cards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 73d96269c9b2..f6e922cf3fd3 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -5125,11 +5125,18 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, autosync_ref = "AES7"; break; case HDSPM_AES32_AUTOSYNC_FROM_AES8: autosync_ref = "AES8"; break; + case HDSPM_AES32_AUTOSYNC_FROM_TCO: + autosync_ref = "TCO"; break; + case HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN: + autosync_ref = "Sync In"; break; default: autosync_ref = "---"; break; } snd_iprintf(buffer, "AutoSync ref = %s\n", autosync_ref); + /* call readout function for TCO specific status */ + snd_hdspm_proc_read_tco(entry, buffer); + snd_iprintf(buffer, "\n"); } -- cgit v1.2.3 From 2336142fc0470db2ac831225936b8e37b3ecb2bd Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:17 +0200 Subject: ALSA: hdspm - Introduce hdspm_external_rate_to_enum() helper function This patch refactors the code to query the external sample rate and its translation into the corresponding enum into a helper function to prevent future code duplication. A later commit will make use of this new helper function. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 28 +++++++++++++++++----------- 1 file changed, 17 insertions(+), 11 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f6e922cf3fd3..26f10fdbfcd6 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2303,6 +2303,21 @@ static void snd_hdspm_set_infotext(struct snd_ctl_elem_info *uinfo, snd_hdspm_set_infotext(info, texts, ARRAY_SIZE(texts)) +/* Helper function to query the external sample rate and return the + * corresponding enum to be returned to userspace. + */ +static int hdspm_external_rate_to_enum(struct hdspm *hdspm) +{ + int rate = hdspm_external_sample_rate(hdspm); + int i, selected_rate = 0; + for (i = 1; i < 10; i++) + if (HDSPM_bit2freq(i) == rate) { + selected_rate = i; + break; + } + return selected_rate; +} + #define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -2396,18 +2411,9 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, case MADI: case MADIface: - { - int rate = hdspm_external_sample_rate(hdspm); - int i, selected_rate = 0; - for (i = 1; i < 10; i++) - if (HDSPM_bit2freq(i) == rate) { - selected_rate = i; - break; - } - ucontrol->value.enumerated.item[0] = selected_rate; - } + ucontrol->value.enumerated.item[0] = + hdspm_external_rate_to_enum(hdspm); break; - default: break; } -- cgit v1.2.3 From 2d63ec38f5bb1f598baa003a964805c852a80b33 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:18 +0200 Subject: ALSA: hdspm - AES32: Report external sample rate to userspace This patch adds a new ALSA control to read the external sample rate from userspace on RME AES(32) cards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 26f10fdbfcd6..2f58e0721ea3 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2401,10 +2401,15 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, ucontrol->value.enumerated.item[0] = hdspm_get_sync_in_sample_rate(hdspm); break; + case 11: /* External Rate */ + ucontrol->value.enumerated.item[0] = + hdspm_external_rate_to_enum(hdspm); + break; default: /* AES1 to AES8 */ ucontrol->value.enumerated.item[0] = - hdspm_get_s1_sample_rate(hdspm, - kcontrol->private_value-1); + hdspm_get_aes_sample_rate(hdspm, + kcontrol->private_value - + HDSPM_AES32_AUTOSYNC_FROM_AES1); break; } break; @@ -4550,7 +4555,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = { HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0), HDSPM_AUTOSYNC_REF("AutoSync Reference", 0), HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0), - HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0), + HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 11), HDSPM_SYNC_CHECK("WC Sync Check", 0), HDSPM_SYNC_CHECK("AES1 Sync Check", 1), HDSPM_SYNC_CHECK("AES2 Sync Check", 2), -- cgit v1.2.3 From 0dc831b9bca98a0d8dafb00fa7f20b3aef6cab67 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:19 +0200 Subject: ALSA: hdspm - AES32: Enable TCO support This patch finally enables TCO support on RME AES(32) cards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 2f58e0721ea3..630316c108c0 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6811,6 +6811,7 @@ static int snd_hdspm_create(struct snd_card *card, break; case MADI: + case AES32: if (hdspm_read(hdspm, HDSPM_statusRegister) & HDSPM_tco_detect) { hdspm->midiPorts++; hdspm->tco = kzalloc(sizeof(struct hdspm_tco), -- cgit v1.2.3 From 38816545a2cc6f436e5f9b26ebcb4cc2813eeb5c Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:20 +0200 Subject: ALSA: hdspm - Use snd_ctl_enum_info for most text arrays Use snd_ctl_enum_info() to fill most of the enumerated controls. More non-trivial occurrences will follow in separate commits. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 50 ++++++++++++++++++----------------------------- 1 file changed, 19 insertions(+), 31 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 630316c108c0..5a2eb644458e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -590,7 +590,7 @@ static char *texts_autosync_aio_tco[] = { static char *texts_autosync_aio[] = { "Word Clock", "ADAT", "AES", "SPDIF", "Sync In" }; -static char *texts_freq[] = { +static const char *const texts_freq[] = { "No Lock", "32 kHz", "44.1 kHz", @@ -2286,21 +2286,8 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx) return (status >> (idx*4)) & 0xF; } -static void snd_hdspm_set_infotext(struct snd_ctl_elem_info *uinfo, - char **texts, const int count) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = count; - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); -} - #define ENUMERATED_CTL_INFO(info, texts) \ - snd_hdspm_set_infotext(info, texts, ARRAY_SIZE(texts)) + snd_ctl_enum_info(info, 1, ARRAY_SIZE(texts), texts) /* Helper function to query the external sample rate and return the @@ -2477,7 +2464,7 @@ static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode) static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Master", "AutoSync" }; + static const char *const texts[] = { "Master", "AutoSync" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3057,7 +3044,7 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_video_input_format(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No video", "NTSC", "PAL"}; + static const char *const texts[] = {"No video", "NTSC", "PAL"}; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3103,7 +3090,7 @@ static int snd_hdspm_get_tco_video_input_format(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_ltc_frames(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps", + static const char *const texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps", "30 fps"}; ENUMERATED_CTL_INFO(uinfo, texts); return 0; @@ -3253,7 +3240,7 @@ static int hdspm_set_input_select(struct hdspm * hdspm, int out) static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "optical", "coaxial" }; + static const char *const texts[] = { "optical", "coaxial" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3315,7 +3302,7 @@ static int hdspm_set_ds_wire(struct hdspm * hdspm, int ds) static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Single", "Double" }; + static const char *const texts[] = { "Single", "Double" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3388,7 +3375,7 @@ static int hdspm_set_qs_wire(struct hdspm * hdspm, int mode) static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Single", "Double", "Quad" }; + static const char *const texts[] = { "Single", "Double", "Quad" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3454,8 +3441,8 @@ static int snd_hdspm_info_tristate(struct snd_kcontrol *kcontrol, { u32 regmask = kcontrol->private_value; - static char *texts_spdif[] = { "Optical", "Coaxial", "Internal" }; - static char *texts_levels[] = { "Hi Gain", "+4 dBu", "-10 dBV" }; + static const char *const texts_spdif[] = { "Optical", "Coaxial", "Internal" }; + static const char *const texts_levels[] = { "Hi Gain", "+4 dBu", "-10 dBV" }; switch (regmask) { case HDSPM_c0_Input0: @@ -3542,7 +3529,7 @@ static int hdspm_set_madi_speedmode(struct hdspm *hdspm, int mode) static int snd_hdspm_info_madi_speedmode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "Single", "Double", "Quad" }; + static const char *const texts[] = { "Single", "Double", "Quad" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3777,7 +3764,7 @@ static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "No Lock", "Lock", "Sync", "N/A" }; + static const char *const texts[] = { "No Lock", "Lock", "Sync", "N/A" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -3785,7 +3772,7 @@ static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol, static int snd_hdspm_tco_info_lock_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "No Lock", "Lock" }; + static const char *const texts[] = { "No Lock", "Lock" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4175,7 +4162,7 @@ static void hdspm_tco_write(struct hdspm *hdspm) static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "44.1 kHz", "48 kHz" }; + static const char *const texts[] = { "44.1 kHz", "48 kHz" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4221,7 +4208,8 @@ static int snd_hdspm_put_tco_sample_rate(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_pull(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "0", "+ 0.1 %", "- 0.1 %", "+ 4 %", "- 4 %" }; + static const char *const texts[] = { "0", "+ 0.1 %", "- 0.1 %", + "+ 4 %", "- 4 %" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4266,7 +4254,7 @@ static int snd_hdspm_put_tco_pull(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_wck_conversion(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" }; + static const char *const texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } @@ -4312,7 +4300,7 @@ static int snd_hdspm_put_tco_wck_conversion(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_frame_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "24 fps", "25 fps", "29.97fps", + static const char *const texts[] = { "24 fps", "25 fps", "29.97fps", "29.97 dfps", "30 fps", "30 dfps" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; @@ -4359,7 +4347,7 @@ static int snd_hdspm_put_tco_frame_rate(struct snd_kcontrol *kcontrol, static int snd_hdspm_info_tco_sync_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = { "LTC", "Video", "WCK" }; + static const char *const texts[] = { "LTC", "Video", "WCK" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; } -- cgit v1.2.3 From eb0d4dbf3d7f503f435022da46ef1495ca570d85 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:21 +0200 Subject: ALSA: hdspm - Use snd_ctl_enum_info() for texts_autosync Also use snd_ctl_enum_info() to fill the autosync enumerated controls. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 29 ++++++++++------------------- 1 file changed, 10 insertions(+), 19 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 5a2eb644458e..ffd5d7c08a4b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -558,36 +558,36 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); /* names for speed modes */ static char *hdspm_speed_names[] = { "single", "double", "quad" }; -static char *texts_autosync_aes_tco[] = { "Word Clock", +static const char *const texts_autosync_aes_tco[] = { "Word Clock", "AES1", "AES2", "AES3", "AES4", "AES5", "AES6", "AES7", "AES8", "TCO", "Sync In" }; -static char *texts_autosync_aes[] = { "Word Clock", +static const char *const texts_autosync_aes[] = { "Word Clock", "AES1", "AES2", "AES3", "AES4", "AES5", "AES6", "AES7", "AES8", "Sync In" }; -static char *texts_autosync_madi_tco[] = { "Word Clock", +static const char *const texts_autosync_madi_tco[] = { "Word Clock", "MADI", "TCO", "Sync In" }; -static char *texts_autosync_madi[] = { "Word Clock", +static const char *const texts_autosync_madi[] = { "Word Clock", "MADI", "Sync In" }; -static char *texts_autosync_raydat_tco[] = { +static const char *const texts_autosync_raydat_tco[] = { "Word Clock", "ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4", "AES", "SPDIF", "TCO", "Sync In" }; -static char *texts_autosync_raydat[] = { +static const char *const texts_autosync_raydat[] = { "Word Clock", "ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4", "AES", "SPDIF", "Sync In" }; -static char *texts_autosync_aio_tco[] = { +static const char *const texts_autosync_aio_tco[] = { "Word Clock", "ADAT", "AES", "SPDIF", "TCO", "Sync In" }; -static char *texts_autosync_aio[] = { "Word Clock", +static const char *const texts_autosync_aio[] = { "Word Clock", "ADAT", "AES", "SPDIF", "Sync In" }; static const char *const texts_freq[] = { @@ -975,7 +975,7 @@ struct hdspm { struct hdspm_tco *tco; /* NULL if no TCO detected */ - char **texts_autosync; + const char *const *texts_autosync; int texts_autosync_items; cycles_t last_interrupt; @@ -2888,16 +2888,7 @@ static int snd_hdspm_info_pref_sync_ref(struct snd_kcontrol *kcontrol, { struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = hdspm->texts_autosync_items; - - if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - - strcpy(uinfo->value.enumerated.name, - hdspm->texts_autosync[uinfo->value.enumerated.item]); + snd_ctl_enum_info(uinfo, 1, hdspm->texts_autosync_items, hdspm->texts_autosync); return 0; } -- cgit v1.2.3 From 04659f9e9e6f2493d0e2dc52c72c4f20c22d9c61 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Fri, 5 Jul 2013 11:28:22 +0200 Subject: ALSA: hdspm - Use snd_ctl_enum_info() in snd_hdspm_info_autosync_ref Also use snd_ctl_enum_info() to fill the autosync text fields on AES32 and MADI cards (only users of snd_hdspm_info_autosync_ref). Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 24 ++++-------------------- 1 file changed, 4 insertions(+), 20 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ffd5d7c08a4b..7a09b2de907b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -2983,31 +2983,15 @@ static int snd_hdspm_info_autosync_ref(struct snd_kcontrol *kcontrol, struct hdspm *hdspm = snd_kcontrol_chip(kcontrol); if (AES32 == hdspm->io_type) { - static char *texts[] = { "WordClock", "AES1", "AES2", "AES3", + static const char *const texts[] = { "WordClock", "AES1", "AES2", "AES3", "AES4", "AES5", "AES6", "AES7", "AES8", "TCO", "Sync In", "None"}; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = ARRAY_SIZE(texts); - if (uinfo->value.enumerated.item >= - uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); + ENUMERATED_CTL_INFO(uinfo, texts); } else if (MADI == hdspm->io_type) { - static char *texts[] = {"Word Clock", "MADI", "TCO", + static const char *const texts[] = {"Word Clock", "MADI", "TCO", "Sync In", "None" }; - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = 5; - if (uinfo->value.enumerated.item >= - uinfo->value.enumerated.items) - uinfo->value.enumerated.item = - uinfo->value.enumerated.items - 1; - strcpy(uinfo->value.enumerated.name, - texts[uinfo->value.enumerated.item]); + ENUMERATED_CTL_INFO(uinfo, texts); } return 0; } -- cgit v1.2.3 From 69358fca4203eda93e008f234fabf603d9dba15e Mon Sep 17 00:00:00 2001 From: Martin Dausel Date: Fri, 5 Jul 2013 11:28:23 +0200 Subject: ALSA: hdspm - Added some comments and control register documentation Signed-off-by: Martin Dausel Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 106 +++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 99 insertions(+), 7 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 7a09b2de907b..a3a71ac513f1 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -38,6 +38,97 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ + +/* ************* Register Documentation ******************************************************* + * + * Work in progress! Documentation is based on the code in this file. + * + * --------- HDSPM_controlRegister --------- + * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :3322.2222:2222.1111:1111.1100:0000.0000: bit number + * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31 + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit + * : . : . : . : x . : HDSPM_AudioInterruptEnable \_ setting both bits + * : . : . : . : . x: HDSPM_Start / enables audio IO + * : . : . : . : x. : HDSPM_ClockModeMaster - 1: Master, 0: Slave + * : . : . : . : .210 : HDSPM_LatencyMask - 3 Bit value for latency + * : . : . : . : . : 0:64, 1:128, 2:256, 3:512, + * : . : . : . : . : 4:1024, 5:2048, 6:4096, 7:8192 + * :x . : . : . x:xx . : HDSPM_FrequencyMask + * : . : . : . :10 . : HDSPM_Frequency1|HDSPM_Frequency0: 1=32K,2=44.1K,3=48K,0=?? + * : . : . : . x: . : HDSPM_DoubleSpeed + * :x . : . : . : . : HDSPM_QuadSpeed + * : . 3 : . 10: 2 . : . : HDSPM_SyncRefMask : + * : . : . x: . : . : HDSPM_SyncRef0 + * : . : . x : . : . : HDSPM_SyncRef1 + * : . : . : x . : . : HDSPM_SyncRef2 + * : . x : . : . : . : HDSPM_SyncRef3 + * : . : . 10: . : . : sync ref: 0:WC, 1:Madi, 2:TCO, 3:SyncIn + * : . 3 : . 10: 2 . : . : 0:WC, 1:AES1 ... 8:AES8, 9: TCO, 10:SyncIn? + * : . x : . : . : . : HDSPe_FLOAT_FORMAT + * : . : . : x . : . : HDSPM_InputSelect0 : 0=optical,1=coax + * : . : . :x . : . : HDSPM_InputSelect1 + * : . : .x : . : . : HDSPM_clr_tms + * : . : . : . x : . : HDSPM_TX_64ch + * : . : . : . x : . : HDSPM_Emphasis + * : . : . : .x : . : HDSPM_AutoInp + * : . : . x : . : . : HDSPM_SMUX + * : . : .x : . : . : HDSPM_clr_tms + * : . : x. : . : . : HDSPM_taxi_reset + * : . x: . : . : . : HDSPM_LineOut + * : . x: . : . : . : ?????????????????? + * : . : x. : . : . : HDSPM_WCK48 + * : . : . : .x : . : HDSPM_Dolby + * : . : x . : . : . : HDSPM_Midi0InterruptEnable + * : . :x . : . : . : HDSPM_Midi1InterruptEnable + * : . : x . : . : . : HDSPM_Midi2InterruptEnable + * : . x : . : . : . : HDSPM_Midi3InterruptEnable + * : . x : . : . : . : HDSPM_DS_DoubleWire + * : .x : . : . : . : HDSPM_QS_DoubleWire + * : x. : . : . : . : HDSPM_QS_QuadWire + * : . : . : . x : . : HDSPM_Professional + * : x . : . : . : . : HDSPM_wclk_sel + * : . : . : . : . : + * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :3322.2222:2222.1111:1111.1100:0000.0000: bit number + * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31 + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :8421.8421:8421.8421:8421.8421:8421.8421:hex digit + * + * + * + * AIO / RayDAT only + * + * ------------ HDSPM_WR_SETTINGS ---------- + * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte + * :1098.7654:3210.9876:5432.1098:7654.3210: + * :||||.||||:||||.||||:||||.||||:||||.||||: bit number + * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31 + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit + * : . : . : . : . x: HDSPM_c0Master 1: Master, 0: Slave + * : . : . : . : . x : HDSPM_c0_SyncRef0 + * : . : . : . : . x : HDSPM_c0_SyncRef1 + * : . : . : . : .x : HDSPM_c0_SyncRef2 + * : . : . : . : x. : HDSPM_c0_SyncRef3 + * : . : . : . : 3.210 : HDSPM_c0_SyncRefMask: + * : . : . : . : . : RayDat: 0:WC, 1:AES, 2:SPDIF, 3..6: ADAT1..4, + * : . : . : . : . : 9:TCO, 10:SyncIn + * : . : . : . : . : AIO: 0:WC, 1:AES, 2: SPDIF, 3: ATAT, + * : . : . : . : . : 9:TCO, 10:SyncIn + * : . : . : . : . : + * : . : . : . : . : + * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte + * :1098.7654:3210.9876:5432.1098:7654.3210: + * :||||.||||:||||.||||:||||.||||:||||.||||: bit number + * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31 + * :||||.||||:||||.||||:||||.||||:||||.||||: + * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit + * + */ #include #include #include @@ -95,7 +186,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_controlRegister 64 #define HDSPM_interruptConfirmation 96 #define HDSPM_control2Reg 256 /* not in specs ???????? */ -#define HDSPM_freqReg 256 /* for AES32 */ +#define HDSPM_freqReg 256 /* for setting arbitrary clock values (DDS feature) */ #define HDSPM_midiDataOut0 352 /* just believe in old code */ #define HDSPM_midiDataOut1 356 #define HDSPM_eeprom_wr 384 /* for AES32 */ @@ -890,11 +981,11 @@ struct hdspm_midi { }; struct hdspm_tco { - int input; - int framerate; - int wordclock; - int samplerate; - int pull; + int input; /* 0: LTC, 1:Video, 2: WC*/ + int framerate; /* 0=24, 1=25, 2=29.97, 3=29.97d, 4=30, 5=30d */ + int wordclock; /* 0=1:1, 1=44.1->48, 2=48->44.1 */ + int samplerate; /* 0=44.1, 1=48, 2= freq from app */ + int pull; /* 0=0, 1=+0.1%, 2=-0.1%, 3=+4%, 4=-4%*/ int term; /* 0 = off, 1 = on */ }; @@ -913,7 +1004,7 @@ struct hdspm { u32 control_register; /* cached value */ u32 control2_register; /* cached value */ - u32 settings_register; + u32 settings_register; /* cached value for AIO / RayDat (sync reference, master/slave) */ struct hdspm_midi midi[4]; struct tasklet_struct midi_tasklet; @@ -4137,6 +4228,7 @@ static void hdspm_tco_write(struct hdspm *hdspm) static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { + /* TODO freq from app could be supported here, see tco->samplerate */ static const char *const texts[] = { "44.1 kHz", "48 kHz" }; ENUMERATED_CTL_INFO(uinfo, texts); return 0; -- cgit v1.2.3 From b6c44f41823e50a5e109e929e07d787eabf4b0d3 Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Wed, 10 Jul 2013 00:22:46 +0900 Subject: ALSA: firewire-speakers: remove not-reused member from structure "pcm" member in struct fwspk is used to set pcm operations but is not used again. This commit remove this member and set pcm operations with snd_pcm_set_ops(). Signed-off-by: Takashi Sakamoto Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/speakers.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index d6846557f270..0f1e5d88ed15 100644 --- a/sound/firewire/speakers.c +++ b/sound/firewire/speakers.c @@ -49,7 +49,6 @@ struct fwspk { struct snd_card *card; struct fw_unit *unit; const struct device_info *device_info; - struct snd_pcm_substream *pcm; struct mutex mutex; struct cmp_connection connection; struct amdtp_out_stream stream; @@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk) return err; pcm->private_data = fwspk; strcpy(pcm->name, fwspk->device_info->short_name); - fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - fwspk->pcm->ops = &ops; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops); return 0; } -- cgit v1.2.3