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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d97f0d6..2ae7dc5 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2ae7dc5..0262ffb 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0262ffb..a667256 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index a667256..876d836 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 876d836..bfa8f53 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index bfa8f53..4fedd9d 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 4fedd9d..7777a3a 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 7777a3a..056810c 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 056810c..1e4dc98 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1e4dc98..3f2434a 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 3f2434a..a41e121 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index a41e121..3b23280 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 v0.10.2 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 diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 3b23280..0cbdd87 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index bd50193..a0fc961 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a0fc961..32a87dc 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 32a87dc..118d727 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 118d727..631c546 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 631c546..4a3a822 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 4a3a822..15f1e7b 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 15f1e7b..b271853 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b271853..d9532c4 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d9532c4..778fc23 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 778fc23..ad41636 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ad41636..06e69de 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 06e69de..58b2104 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 58b2104..bdd8c77 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index bdd8c77..d95100e 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d95100e..d1e0582 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d1e0582..8e6ce14 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 8e6ce14..b7702b2 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index b7702b2..367dd41 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 367dd41..a69957c 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a69957c..c0143cf 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index c0143cf..a9f4c7c 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a9f4c7c..80b2247 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 80b2247..73d9626 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 73d9626..f6e922c 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index f6e922c..26f10fd 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 26f10fd..2f58e07 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 2f58e07..630316c 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 630316c..5a2eb64 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 5a2eb64..ffd5d7c 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ffd5d7c..7a09b2d 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 v0.10.2 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 diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 7a09b2d..a3a71ac 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 v0.10.2 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 diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c index d684655..0f1e5d8 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 v0.10.2 From 68593c9340847662ac1d337b3c5621a1f4ca05db Mon Sep 17 00:00:00 2001 From: Fengguang Wu Date: Mon, 15 Jul 2013 21:41:32 +0800 Subject: ALSA: hdspm - remove unneeded semicolon sound/pci/rme9652/hdspm.c:1110:2-3: Unneeded semicolon Generated by: coccinelle/misc/semicolon.cocci Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a3a71ac..ec6335e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1250,7 +1250,7 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate) else if (hdspm->control_register & HDSPM_DoubleSpeed) return rate * 2; - }; + } return rate; } -- cgit v0.10.2 From e4c3bce26de240457370d00ce396602cc98bb3cc Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 16 Jul 2013 11:48:10 +0200 Subject: ALSA: hda - Headphone mic support for an Asus/Conexant device This Conexant codec has a single jack that can be used as either headphone or mic (but not headset). The existing hp_mic functionality does not apply here, because the mic and the HP are on separate pins. Hence make a lighter version of what has been earlier done for Realtek codecs. BugLink: https://bugs.launchpad.net/bugs/1198030 Tested-by: Franz Hsieh Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index de00ce1..4edd2d0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -66,6 +66,8 @@ struct conexant_spec { hda_nid_t eapds[4]; bool dynamic_eapd; + unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */ + #ifdef ENABLE_CXT_STATIC_QUIRKS const struct snd_kcontrol_new *mixers[5]; int num_mixers; @@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec) snd_hda_gen_init(codec); if (!spec->dynamic_eapd) cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true); + + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); + return 0; } @@ -3224,6 +3229,8 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, CXT_FIXUP_INC_MIC_BOOST, + CXT_FIXUP_HEADPHONE_MIC_PIN, + CXT_FIXUP_HEADPHONE_MIC, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec, (0 << AC_AMPCAP_MUTE_SHIFT)); } +static void cxt_update_headset_mode(struct hda_codec *codec) +{ + /* The verbs used in this function were tested on a Conexant CX20751/2 codec. */ + int i; + bool mic_mode = false; + struct conexant_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + + hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]]; + + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].pin == mux_pin) { + mic_mode = !!cfg->inputs[i].is_headphone_mic; + break; + } + + if (mic_mode) { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */ + spec->gen.hp_jack_present = false; + } else { + snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */ + spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]); + } + + snd_hda_gen_update_outputs(codec); +} + +static void cxt_update_headset_mode_hook(struct hda_codec *codec, + struct snd_ctl_elem_value *ucontrol) +{ + cxt_update_headset_mode(codec); +} + +static void cxt_fixup_headphone_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct conexant_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC; + break; + case HDA_FIXUP_ACT_PROBE: + spec->gen.cap_sync_hook = cxt_update_headset_mode_hook; + spec->gen.automute_hook = cxt_update_headset_mode; + break; + case HDA_FIXUP_ACT_INIT: + cxt_update_headset_mode(codec); + break; + } +} + + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, }, + [CXT_FIXUP_HEADPHONE_MIC_PIN] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_HEADPHONE_MIC, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */ + { } + } + }, + [CXT_FIXUP_HEADPHONE_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt_fixup_headphone_mic, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), @@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec) snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); - err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0); + err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, + spec->parse_flags); if (err < 0) goto error; @@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec) codec->bus->allow_bus_reset = 1; } + snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); + return 0; error: -- cgit v0.10.2 From 60ea8ca21b4584cebb8163879b50ab3d941090bf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Jul 2013 16:59:46 +0200 Subject: ALSA: hda - Add snd_hda_jack_detect_state() helper function snd_hda_jack_detect() function returns a boolean value for a jack plugged in or not, but it also returns always true when the corresponding pin is phantom (i.e. fixed). This is OK in most cases, but it makes the generic parser misbehaving about the auto-mute or auto-mic switching, e.g. when one of headphone pins is a fixed. Namely, the driver decides whether to mute the speaker or not, just depending on the headphone plug state: if one of the headphone jacks is seen as active, then the speaker is muted. Thus this will result always in the muted speaker output. So, the problem is the function returns a boolean, after all, although we need to think of "phantom" jack. Now a new function, snd_hda_jack_detect_state() is introduced to return these tristates. The generic parser uses this function for checking the headphone or mic jack states. Meanwhile, the behavior of snd_hda_jack_detect() is kept as is, for keeping compatibility in other driver codes. Acked-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbb..f5c2d1f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3724,7 +3724,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, /* check each pin in the given array; returns true if any of them is plugged */ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) { - int i, present = 0; + int i; + bool present = false; for (i = 0; i < num_pins; i++) { hda_nid_t nid = pins[i]; @@ -3733,7 +3734,8 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) /* don't detect pins retasked as inputs */ if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN) continue; - present |= snd_hda_jack_detect(codec, nid); + if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT) + present = true; } return present; } @@ -3887,7 +3889,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; - if (snd_hda_jack_detect(codec, pin)) { + if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return; } diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 3fd2973..dc93761 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid) EXPORT_SYMBOL_HDA(snd_hda_pin_sense); /** - * snd_hda_jack_detect - query pin Presence Detect status + * snd_hda_jack_detect_state - query pin Presence Detect status * @codec: the CODEC to sense * @nid: the pin NID to sense * - * Query and return the pin's Presence Detect status. + * Query and return the pin's Presence Detect status, as either + * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM. */ -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid) { - u32 sense = snd_hda_pin_sense(codec, nid); - return get_jack_plug_state(sense); + struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid); + if (jack && jack->phantom_jack) + return HDA_JACK_PHANTOM; + else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE) + return HDA_JACK_PRESENT; + else + return HDA_JACK_NOT_PRESENT; } -EXPORT_SYMBOL_HDA(snd_hda_jack_detect); +EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state); /** * snd_hda_jack_detect_enable - enable the jack-detection diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h index ec12abd..379420c 100644 --- a/sound/pci/hda/hda_jack.h +++ b/sound/pci/hda/hda_jack.h @@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); -int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); + +/* the jack state returned from snd_hda_jack_detect_state() */ +enum { + HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM, +}; + +int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid); + +static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid) +{ + return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT; +} bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid); -- cgit v0.10.2 From b785a492c6eef578520594d5c4d6e9f2cb47cbeb Mon Sep 17 00:00:00 2001 From: Jingoo Han Date: Fri, 19 Jul 2013 16:24:59 +0900 Subject: ALSA: replace strict_strto*() with kstrto*() The usage of strict_strto*() is not preferred, because strict_strto*() is obsolete. Thus, kstrto*() should be used. Signed-off-by: Jingoo Han Signed-off-by: Takashi Iwai diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 11048cc..915b4d7 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry, if (i >= ARRAY_SIZE(fields)) continue; snd_info_get_str(item, ptr, sizeof(item)); - if (strict_strtoull(item, 0, &val)) + if (kstrtoull(item, 0, &val)) continue; if (fields[i].size == sizeof(int)) *get_dummy_int_ptr(dummy, fields[i].offset) = val; diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index ce67608..fe0bda1 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev, \ struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ struct hda_codec *codec = hwdep->private_data; \ unsigned long val; \ - int err = strict_strtoul(buf, 0, &val); \ + int err = kstrtoul(buf, 0, &val); \ if (err < 0) \ return err; \ codec->type = val; \ @@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp) p = snd_hda_get_hint(codec, key); if (!p) ret = -ENOENT; - else if (strict_strtoul(p, 0, &val)) + else if (kstrtoul(p, 0, &val)) ret = -EINVAL; else { *valp = val; @@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \ struct hda_codec **codecp) \ { \ unsigned long val; \ - if (!strict_strtoul(buf, 0, &val)) \ + if (!kstrtoul(buf, 0, &val)) \ (*codecp)->name = val; \ } diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index e2de9ec..e37c06f 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3175,7 +3175,7 @@ static ssize_t wm8962_beep_set(struct device *dev, long int time; int ret; - ret = strict_strtol(buf, 10, &time); + ret = kstrtol(buf, 10, &time); if (ret != 0) return ret; diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index eb68c7d..e4980c5 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev, \ unsigned long val; \ int status; \ \ - status = strict_strtoul(buf, 0, &val); \ + status = kstrtoul(buf, 0, &val); \ if (status) \ return status; \ \ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0ec070c..88daa64 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -192,7 +192,7 @@ static ssize_t pmdown_time_set(struct device *dev, struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev); int ret; - ret = strict_strtol(buf, 10, &rtd->pmdown_time); + ret = kstrtol(buf, 10, &rtd->pmdown_time); if (ret) return ret; @@ -237,6 +237,7 @@ static ssize_t codec_reg_write_file(struct file *file, char *start = buf; unsigned long reg, value; struct snd_soc_codec *codec = file->private_data; + int ret; buf_size = min(count, (sizeof(buf)-1)); if (copy_from_user(buf, user_buf, buf_size)) @@ -248,8 +249,9 @@ static ssize_t codec_reg_write_file(struct file *file, reg = simple_strtoul(start, &start, 16); while (*start == ' ') start++; - if (strict_strtoul(start, 16, &value)) - return -EINVAL; + ret = kstrtoul(start, 16, &value); + if (ret) + return ret; /* Userspace has been fiddling around behind the kernel's back */ add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE); -- cgit v0.10.2 From 9bfb2844a2f9e6eab52aed1eca0d03f4398c755f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Jul 2013 14:31:50 +0200 Subject: ALSA: hda/realtek - Selectively call snd_hda_shutup_pins() Instead of calling snd_hda_shutup_pins() unconditionally, allow it be called in spec->shutup callback. In this way, we can avoid calling this function if it causes a problem like we see in the next patch following this. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd2261..dbd59df 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -282,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec) { alc_auto_setup_eapd(codec, false); msleep(200); + snd_hda_shutup_pins(codec); } /* generic EAPD initialization */ @@ -826,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec) if (spec && spec->shutup) spec->shutup(codec); - snd_hda_shutup_pins(codec); + else + snd_hda_shutup_pins(codec); } #define alc_free snd_hda_gen_free @@ -2573,15 +2575,13 @@ static void alc269_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - if (spec->codec_variant != ALC269_TYPE_ALC269VB) - return; - if (spec->codec_variant == ALC269_TYPE_ALC269VB) alc269vb_toggle_power_output(codec, 0); if (spec->codec_variant == ALC269_TYPE_ALC269VB && (alc_get_coef0(codec) & 0x00ff) == 0x018) { msleep(150); } + snd_hda_shutup_pins(codec); } static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, -- cgit v0.10.2 From c5177c861e2bae584996f60667dc7b291ba6600a Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Wed, 24 Jul 2013 14:39:49 +0200 Subject: ALSA: hda - Fix the noise after suspend on ALC283 codec When the power state of ALC283 codec goes to D3, it gives a noise via headphone output. This is because the driver tries to clear all pins via snd_hda_shutup_pins(). Setting the mic pin to zero triggers such a noise. Define a new shutup call specific to this codec and control the pins there more precisely. Also, add the power-save enable/disable sequences in the resume and the new shutup calls. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dbd59df..04a69e3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2712,6 +2712,13 @@ static int alc269_resume(struct hda_codec *codec) hda_call_check_power_status(codec, 0x01); if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); + + /* clear the power-save mode for ALC283 */ + if (codec->vendor_id == 0x10ec0283) { + alc_write_coef_idx(codec, 0x4, 0xaf01); + alc_write_coef_idx(codec, 0x6, 0x2104); + } + return 0; } #endif /* CONFIG_PM */ @@ -3775,6 +3782,30 @@ static void alc269_fill_coef(struct hda_codec *codec) alc_write_coef_idx(codec, 0x4, val | (1<<11)); } +/* don't clear mic pin; otherwise it results in noise in D3 */ +static void alc283_headset_shutup(struct hda_codec *codec) +{ + int i; + + if (codec->bus->shutdown) + return; + + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + /* use read here for syncing after issuing each verb */ + if (pin->nid != 0x19) + snd_hda_codec_read(codec, pin->nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } + + alc_write_coef_idx(codec, 0x4, 0x0f01); /* power save */ + alc_write_coef_idx(codec, 0x6, 0x2100); /* power save */ + snd_hda_codec_write(codec, 0x19, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + PIN_VREFHIZ); + codec->pins_shutup = 1; +} + /* */ static int patch_alc269(struct hda_codec *codec) @@ -3789,6 +3820,9 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; + if (codec->vendor_id == 0x10ec0283) + spec->shutup = alc283_headset_shutup; + snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -3862,7 +3896,8 @@ static int patch_alc269(struct hda_codec *codec) codec->patch_ops.suspend = alc269_suspend; codec->patch_ops.resume = alc269_resume; #endif - spec->shutup = alc269_shutup; + if (!spec->shutup) + spec->shutup = alc269_shutup; snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE); -- cgit v0.10.2 From da7db6ad4da05a3109d0a31100e1ecd746a90fee Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Mon, 22 Jul 2013 03:19:18 -0400 Subject: ALSA: hda - use azx_writew() for 16-bit length register Register STATESTS is 16-bit length, use correct API for read/write. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8860dd5..3f16c4b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1160,7 +1160,7 @@ static int azx_reset(struct azx *chip, int full_reset) goto __skip; /* clear STATESTS */ - azx_writeb(chip, STATESTS, STATESTS_INT_MASK); + azx_writew(chip, STATESTS, STATESTS_INT_MASK); /* reset controller */ azx_enter_link_reset(chip); @@ -1242,7 +1242,7 @@ static void azx_int_clear(struct azx *chip) } /* clear STATESTS */ - azx_writeb(chip, STATESTS, STATESTS_INT_MASK); + azx_writew(chip, STATESTS, STATESTS_INT_MASK); /* clear rirb status */ azx_writeb(chip, RIRBSTS, RIRB_INT_MASK); @@ -1451,8 +1451,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) #if 0 /* clear state status int */ - if (azx_readb(chip, STATESTS) & 0x04) - azx_writeb(chip, STATESTS, 0x04); + if (azx_readw(chip, STATESTS) & 0x04) + azx_writew(chip, STATESTS, 0x04); #endif spin_unlock(&chip->reg_lock); -- cgit v0.10.2 From 85054b2153f18eac16df9ff88913c98adea6a23e Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sun, 28 Jul 2013 23:27:38 +0300 Subject: ALSA: usx2y: remove an unneeded check The test here is always true because S[i].urb is an array not a pointer. Also it's bogus because the intent was to test: if (S->urb[i]) { instead of: if (S[i].urb) { Anyway, usb_kill_urb() and usb_free_urb() accept NULL pointers so we can just remove this. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 1f9bbd5..5a51b18 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -305,11 +305,9 @@ static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S) { int i; for (i = 0; i < URBS_AsyncSeq; ++i) { - if (S[i].urb) { - usb_kill_urb(S->urb[i]); - usb_free_urb(S->urb[i]); - S->urb[i] = NULL; - } + usb_kill_urb(S->urb[i]); + usb_free_urb(S->urb[i]); + S->urb[i] = NULL; } kfree(S->buffer); } -- cgit v0.10.2 From 7eaa9161edd1bb41c026db252bb7e7dfe97ab90a Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Thu, 25 Jul 2013 23:34:44 -0400 Subject: ALSA: hda - Clearing jackpoll_interval avoid pending work Clearing jackpoll_interval before calling cancel_delayed_work_sync(), otherwise the work will be triggered again and cause impact in hda_jackpoll_work(). The next patch will poll jack once even with jackpoll_interval=0. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e2481ba..0bc20ef 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec) return; if (spec->hp_work_active) { snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1); + codec->jackpoll_interval = 0; cancel_delayed_work_sync(&codec->jackpoll_work); spec->hp_work_active = false; - codec->jackpoll_interval = 0; } } -- cgit v0.10.2 From 18e606275691726cce06ad803072ac54315740f7 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Thu, 25 Jul 2013 23:34:45 -0400 Subject: ALSA: hda - jack poll once if jackpoll_interval==0 With jackpoll_interval != 0, it's used to poll jack event periodically in a delayed work. if it's 0, give the caller chance to probe jack status but will not restart the delayed work. In the next patch which enable WAKEEN feature, HDA controller was able to wake up system when it's in D3, it's useful to detect Jack hotplug event and notify userspace. By default the jackpoll_interval=0, this patch let jack poll once without starting the delayed work. Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8a005f0..fdbb09a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1216,11 +1216,13 @@ static void hda_jackpoll_work(struct work_struct *work) { struct hda_codec *codec = container_of(work, struct hda_codec, jackpoll_work.work); - if (!codec->jackpoll_interval) - return; snd_hda_jack_set_dirty_all(codec); snd_hda_jack_poll_all(codec); + + if (!codec->jackpoll_interval) + return; + queue_delayed_work(codec->bus->workq, &codec->jackpoll_work, codec->jackpoll_interval); } -- cgit v0.10.2 From 7d4f606c50ffaaa3ac60b7faf770dc6e84af3207 Mon Sep 17 00:00:00 2001 From: Wang Xingchao Date: Thu, 25 Jul 2013 23:34:46 -0400 Subject: ALSA: hda - WAKEEN feature enabling for runtime pm With runtime power save feature enabled, Headphone hotplug event will not be detected while controller/codec in D3. HDA has feature WAKEEN to let codec wake up system if controller is in D3 or system in S3.(HDA Spec 4.5.9.2/3). Codec can send out INT or wake up controller depending on whether CIE or GIE enabled.(Figure 4, Interupt structure). The controller must be in RESET mode after enter runtime-suspend, otherwise it will not be waken up even if codec send out wake-up event. And STATESTS will be cleared after controller brought out of RESET mode. This patch only enable WAKEEN for runtime-suspend(Controller D3) mode, not for system S3 mode. with tool "evtest", Headphone hotplug events could be cought and reported successfully. [fixed an unused variable warning by tiwai] Signed-off-by: Wang Xingchao Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3f16c4b..7f9e406 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2971,6 +2971,10 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + azx_stop_chip(chip); azx_enter_link_reset(chip); azx_clear_irq_pending(chip); @@ -2983,11 +2987,31 @@ static int azx_runtime_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; + struct hda_bus *bus; + struct hda_codec *codec; + int status; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(true); + + /* Read STATESTS before controller reset */ + status = azx_readw(chip, STATESTS); + azx_init_pci(chip); azx_init_chip(chip, 1); + + bus = chip->bus; + if (status && bus) { + list_for_each_entry(codec, &bus->codec_list, list) + if (status & (1 << codec->addr)) + queue_delayed_work(codec->bus->workq, + &codec->jackpoll_work, codec->jackpoll_interval); + } + + /* disable controller Wake Up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & + ~STATESTS_INT_MASK); + return 0; } -- cgit v0.10.2 From eefb8be4a4fb4aa9005fc092a88d66fe7cf1adc2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jul 2013 16:26:15 +0200 Subject: ALSA: hda - Remove analog mic pin override from STAC9228 dell-bios quirk The current fixup for dell-bios model with STAC9228 codec contains the override of pin 0x0c for analog mic. But this is actually just adding a bogus pin and confuses the parser. Better to remove it for the auto-mic switching. Meanwhile, for a possible regression, keep the old configuration as model=dell-bios-amic, so that people can test it again quickly. Tested on Dell 1420n laptop. Reported-and-tested-by: Eric Shattow Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 809d72b..a46ddb8 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -244,6 +244,7 @@ STAC9227/9228/9229/927x 5stack-no-fp D965 5stack without front panel dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup + dell-bios-amic Fixes with Dell BIOS setup including analog mic volknob Fixes with volume-knob widget 0x24 auto BIOS setup (default) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e2f8359..8f6c357 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -158,6 +158,7 @@ enum { STAC_D965_VERBS, STAC_DELL_3ST, STAC_DELL_BIOS, + STAC_DELL_BIOS_AMIC, STAC_DELL_BIOS_SPDIF, STAC_927X_DELL_DMIC, STAC_927X_VOLKNOB, @@ -3228,8 +3229,6 @@ static const struct hda_fixup stac927x_fixups[] = { [STAC_DELL_BIOS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - /* configure the analog microphone on some laptops */ - { 0x0c, 0x90a79130 }, /* correct the front output jack as a hp out */ { 0x0f, 0x0227011f }, /* correct the front input jack as a mic */ @@ -3239,6 +3238,16 @@ static const struct hda_fixup stac927x_fixups[] = { .chained = true, .chain_id = STAC_927X_DELL_DMIC, }, + [STAC_DELL_BIOS_AMIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + /* configure the analog microphone on some laptops */ + { 0x0c, 0x90a79130 }, + {} + }, + .chained = true, + .chain_id = STAC_DELL_BIOS, + }, [STAC_DELL_BIOS_SPDIF] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -3267,6 +3276,7 @@ static const struct hda_model_fixup stac927x_models[] = { { .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" }, { .id = STAC_DELL_3ST, .name = "dell-3stack" }, { .id = STAC_DELL_BIOS, .name = "dell-bios" }, + { .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" }, { .id = STAC_927X_VOLKNOB, .name = "volknob" }, {} }; -- cgit v0.10.2 From da96fb5b0185d27faab0746f872d22b0cee7b026 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jul 2013 16:54:36 +0200 Subject: ALSA: hda - Fix invalid multi-io creation on VAIO-Z laptops VAIO-Z laptops need to use the specific DAC for the speaker output by some unknown reason although the codec itself supports the flexible connection. So we implemented a workaround by a new flag, no_primary_hp, for assigning the speaker pin first. This worked until 3.8 kernel, but it got broken because the driver learned for a better multi-io pin mapping, and not it can assign two mic pins for multi-io. Since the multi-io requires to be the primary output, the hp and two mic pins are assigned in prior to the speaker in the end. Although the machine has two mic pins, one of them is used as a noise- canceling headphone, thus it's no real retaskable mic jack. Thus, at best, we can disable the multi-io assignment and make the parser behavior back to the state before the multi-io. This patch adds again a new flag, no_multi_io, to indicate that the device has no multi-io capability, and set it in the fixup for VAIO-Z. The no_multi_io flag itself can be used generically, added via a helper line, too. Reported-by: Tormen Reported-by: Adam Williamson Signed-off-by: Takashi Iwai diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index c3c912d..42a0a39 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -454,6 +454,8 @@ The generic parser supports the following hints: - need_dac_fix (bool): limits the DACs depending on the channel count - primary_hp (bool): probe headphone jacks as the primary outputs; default true +- multi_io (bool): try probing multi-I/O config (e.g. shared + line-in/surround, mic/clfe jacks) - multi_cap_vol (bool): provide multiple capture volumes - inv_dmic_split (bool): provide split internal mic volume/switch for phase-inverted digital mics diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f5c2d1f..f6c0344 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -142,6 +142,9 @@ static void parse_user_hints(struct hda_codec *codec) val = snd_hda_get_bool_hint(codec, "primary_hp"); if (val >= 0) spec->no_primary_hp = !val; + val = snd_hda_get_bool_hint(codec, "multi_io"); + if (val >= 0) + spec->no_multi_io = !val; val = snd_hda_get_bool_hint(codec, "multi_cap_vol"); if (val >= 0) spec->multi_cap_vol = !!val; @@ -1541,7 +1544,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec, cfg->speaker_pins, spec->multiout.extra_out_nid, spec->speaker_paths); - if (fill_mio_first && cfg->line_outs == 1 && + if (!spec->no_multi_io && + fill_mio_first && cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = fill_multi_ios(codec, cfg->line_out_pins[0], true); if (!err) @@ -1554,7 +1558,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec, spec->private_dac_nids, spec->out_paths, spec->main_out_badness); - if (fill_mio_first && + if (!spec->no_multi_io && fill_mio_first && cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { /* try to fill multi-io first */ err = fill_multi_ios(codec, cfg->line_out_pins[0], false); @@ -1582,7 +1586,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec, return err; badness += err; } - if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { + if (!spec->no_multi_io && + cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = fill_multi_ios(codec, cfg->line_out_pins[0], false); if (err < 0) return err; @@ -1600,7 +1605,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec, check_aamix_out_path(codec, spec->speaker_paths[0]); } - if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) + if (!spec->no_multi_io && + cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) if (count_multiio_pins(codec, cfg->hp_pins[0]) >= 2) spec->multi_ios = 1; /* give badness */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index e199a85..48d4402 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -220,6 +220,7 @@ struct hda_gen_spec { unsigned int hp_mic:1; /* Allow HP as a mic-in */ unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */ unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */ + unsigned int no_multi_io:1; /* Don't try multi I/O config */ unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */ unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */ unsigned int own_eapd_ctl:1; /* set EAPD by own function */ diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 04a69e3..ad7a098 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1845,8 +1845,10 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, const struct hda_fixup *fix, int action) { struct alc_spec *spec = codec->spec; - if (action == HDA_FIXUP_ACT_PRE_PROBE) + if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->gen.no_primary_hp = 1; + spec->gen.no_multi_io = 1; + } } static const struct hda_fixup alc882_fixups[] = { -- cgit v0.10.2 From bde7bc6014a0a6f63cff42211ccd9b7129ce2df9 Mon Sep 17 00:00:00 2001 From: Chih-Chung Chang Date: Mon, 5 Aug 2013 16:38:42 +0800 Subject: ALSA: hda - Fix jack gating when auto_{mute,mic} is suppressed. The snd_hda_jack_set_gating_jack() call didn't work when auto_{mute,mic} is suppressed because (1) am_entry is not filled with nid of the mic pin. (2) The jacks are not created (by snd_hda_jack_detect_enable_callback) before the snd_hda_jack_set_gating_jack call. Now we use the first input pin nid directly, and create the jack if it doesn't exist yet. Signed-off-by: Chih-Chung Chang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index dc93761..05b3e3e 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -253,8 +253,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable); int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid, hda_nid_t gating_nid) { - struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid); - struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid); + struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid); + struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid); if (!gated || !gating) return -EINVAL; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ad7a098..6ac4810 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3260,6 +3260,28 @@ static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, alc_fixup_headset_mode(codec, fix, action); } +/* Returns the nid of the external mic input pin, or 0 if it cannot be found. */ +static int find_ext_mic_pin(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->gen.autocfg; + hda_nid_t nid; + unsigned int defcfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + if (cfg->inputs[i].type != AUTO_PIN_MIC) + continue; + nid = cfg->inputs[i].pin; + defcfg = snd_hda_codec_get_pincfg(codec, nid); + if (snd_hda_get_input_pin_attr(defcfg) == INPUT_PIN_ATTR_INT) + continue; + return nid; + } + + return 0; +} + static void alc271_hp_gate_mic_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -3267,11 +3289,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, struct alc_spec *spec = codec->spec; if (action == HDA_FIXUP_ACT_PROBE) { - if (snd_BUG_ON(!spec->gen.am_entry[1].pin || - !spec->gen.autocfg.hp_pins[0])) + int mic_pin = find_ext_mic_pin(codec); + int hp_pin = spec->gen.autocfg.hp_pins[0]; + + if (snd_BUG_ON(!mic_pin || !hp_pin)) return; - snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin, - spec->gen.autocfg.hp_pins[0]); + snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin); } } -- cgit v0.10.2 From d833cdb10cb689ffcbebbf4bae5227072c53f88a Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sat, 3 Aug 2013 10:50:15 +0200 Subject: ALSA: usb-audio: remove disabled debug code in set_format Code block does not compile when enabled. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 15b151e..3d3e8d1 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -486,15 +486,6 @@ add_sync_ep: snd_usb_set_format_quirk(subs, fmt); -#if 0 - printk(KERN_DEBUG - "setting done: format = %d, rate = %d..%d, channels = %d\n", - fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels); - printk(KERN_DEBUG - " datapipe = 0x%0x, syncpipe = 0x%0x\n", - subs->datapipe, subs->syncpipe); -#endif - return 0; } -- cgit v0.10.2 From d133f2c22e9cb7b6afd170437cf0ef1e8a1571b6 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sat, 3 Aug 2013 10:50:16 +0200 Subject: ALSA: usb-audio: remove assignment from if condition Following general kernel style. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 3d3e8d1..be5c7c2 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -479,7 +479,8 @@ add_sync_ep: subs->data_endpoint->sync_master = subs->sync_endpoint; } - if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0) + err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt); + if (err < 0) return err; subs->cur_audiofmt = fmt; -- cgit v0.10.2 From 71bb64c56d787a221752b1de034fe8c07c737f5c Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sat, 3 Aug 2013 10:50:17 +0200 Subject: ALSA: usb-audio: separate sync endpoint setting from set_format Setting the sync endpoint currently takes up about half of set_format(). Move it to a dedicated function. No functional change. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index be5c7c2..e24ce7d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -327,64 +327,17 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, return 0; } -/* - * find a matching format and set up the interface - */ -static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) + +static int set_sync_endpoint(struct snd_usb_substream *subs, + struct audioformat *fmt, + struct usb_device *dev, + struct usb_host_interface *alts, + struct usb_interface_descriptor *altsd) { - struct usb_device *dev = subs->dev; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; struct usb_interface *iface; - unsigned int ep, attr; int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - int err, implicit_fb = 0; - - iface = usb_ifnum_to_if(dev, fmt->iface); - if (WARN_ON(!iface)) - return -EINVAL; - alts = &iface->altsetting[fmt->altset_idx]; - altsd = get_iface_desc(alts); - if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting)) - return -EINVAL; - - if (fmt == subs->cur_audiofmt) - return 0; - - /* close the old interface */ - if (subs->interface >= 0 && subs->interface != fmt->iface) { - err = usb_set_interface(subs->dev, subs->interface, 0); - if (err < 0) { - snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed (%d)\n", - dev->devnum, fmt->iface, fmt->altsetting, err); - return -EIO; - } - subs->interface = -1; - subs->altset_idx = 0; - } - - /* set interface */ - if (subs->interface != fmt->iface || - subs->altset_idx != fmt->altset_idx) { - err = usb_set_interface(dev, fmt->iface, fmt->altsetting); - if (err < 0) { - snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed (%d)\n", - dev->devnum, fmt->iface, fmt->altsetting, err); - return -EIO; - } - snd_printdd(KERN_INFO "setting usb interface %d:%d\n", - fmt->iface, fmt->altsetting); - subs->interface = fmt->iface; - subs->altset_idx = fmt->altset_idx; - - snd_usb_set_interface_quirk(dev); - } - - subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, - alts, fmt->endpoint, subs->direction, - SND_USB_ENDPOINT_TYPE_DATA); - if (!subs->data_endpoint) - return -EINVAL; + unsigned int ep, attr; + int implicit_fb = 0; /* we need a sync pipe in async OUT or adaptive IN mode */ /* check the number of EP, since some devices have broken @@ -479,6 +432,71 @@ add_sync_ep: subs->data_endpoint->sync_master = subs->sync_endpoint; } + return 0; +} + +/* + * find a matching format and set up the interface + */ +static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) +{ + struct usb_device *dev = subs->dev; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_interface *iface; + int err; + + iface = usb_ifnum_to_if(dev, fmt->iface); + if (WARN_ON(!iface)) + return -EINVAL; + alts = &iface->altsetting[fmt->altset_idx]; + altsd = get_iface_desc(alts); + if (WARN_ON(altsd->bAlternateSetting != fmt->altsetting)) + return -EINVAL; + + if (fmt == subs->cur_audiofmt) + return 0; + + /* close the old interface */ + if (subs->interface >= 0 && subs->interface != fmt->iface) { + err = usb_set_interface(subs->dev, subs->interface, 0); + if (err < 0) { + snd_printk(KERN_ERR "%d:%d:%d: return to setting 0 failed (%d)\n", + dev->devnum, fmt->iface, fmt->altsetting, err); + return -EIO; + } + subs->interface = -1; + subs->altset_idx = 0; + } + + /* set interface */ + if (subs->interface != fmt->iface || + subs->altset_idx != fmt->altset_idx) { + err = usb_set_interface(dev, fmt->iface, fmt->altsetting); + if (err < 0) { + snd_printk(KERN_ERR "%d:%d:%d: usb_set_interface failed (%d)\n", + dev->devnum, fmt->iface, fmt->altsetting, err); + return -EIO; + } + snd_printdd(KERN_INFO "setting usb interface %d:%d\n", + fmt->iface, fmt->altsetting); + subs->interface = fmt->iface; + subs->altset_idx = fmt->altset_idx; + + snd_usb_set_interface_quirk(dev); + } + + subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, fmt->endpoint, subs->direction, + SND_USB_ENDPOINT_TYPE_DATA); + + if (!subs->data_endpoint) + return -EINVAL; + + err = set_sync_endpoint(subs, fmt, dev, alts, altsd); + if (err < 0) + return err; + err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt); if (err < 0) return err; -- cgit v0.10.2 From a60945fd08e45fceca9e3525d70e080f7ad60a4e Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sat, 3 Aug 2013 10:50:18 +0200 Subject: ALSA: usb-audio: move implicit fb quirks to separate function Separate setting implicit feedback quirks from setting a sync endpoint (which may also be explicit feedback or async). Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e24ce7d..0016f28 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -327,24 +327,16 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum, return 0; } - -static int set_sync_endpoint(struct snd_usb_substream *subs, - struct audioformat *fmt, - struct usb_device *dev, - struct usb_host_interface *alts, - struct usb_interface_descriptor *altsd) +static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, + struct usb_device *dev, + struct usb_interface_descriptor *altsd, + unsigned int attr) { + struct usb_host_interface *alts; struct usb_interface *iface; int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int ep, attr; int implicit_fb = 0; - - /* we need a sync pipe in async OUT or adaptive IN mode */ - /* check the number of EP, since some devices have broken - * descriptors which fool us. if it has only one EP, - * assume it as adaptive-out or sync-in. - */ - attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + unsigned int ep; switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ @@ -388,6 +380,45 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, goto add_sync_ep; } + /* No quirk */ + return 0; + +add_sync_ep: + subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, ep, !subs->direction, + implicit_fb ? + SND_USB_ENDPOINT_TYPE_DATA : + SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) + return -EINVAL; + + subs->data_endpoint->sync_master = subs->sync_endpoint; + + return 0; +} + +static int set_sync_endpoint(struct snd_usb_substream *subs, + struct audioformat *fmt, + struct usb_device *dev, + struct usb_host_interface *alts, + struct usb_interface_descriptor *altsd) +{ + int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; + unsigned int ep, attr; + int implicit_fb = 0; + int err; + + /* we need a sync pipe in async OUT or adaptive IN mode */ + /* check the number of EP, since some devices have broken + * descriptors which fool us. if it has only one EP, + * assume it as adaptive-out or sync-in. + */ + attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + + err = set_sync_ep_implicit_fb_quirk(subs, dev, altsd, attr); + if (err < 0) + return err; + if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && altsd->bNumEndpoints >= 2) { @@ -420,7 +451,6 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK) == USB_ENDPOINT_USAGE_IMPLICIT_FB; -add_sync_ep: subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, alts, ep, !subs->direction, implicit_fb ? -- cgit v0.10.2 From f34d0650133389c76e22e9f27e57b74ed9e2c042 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sat, 3 Aug 2013 10:50:19 +0200 Subject: ALSA: usb-audio: reverse condition logic in set_sync_endpoint Reverse logic on the conditions required to qualify for a sync endpoint and remove one level of indendation. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 0016f28..c31dbdc 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -419,49 +419,52 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, if (err < 0) return err; - if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || - (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && - altsd->bNumEndpoints >= 2) { - /* check sync-pipe endpoint */ - /* ... and check descriptor size before accessing bSynchAddress - because there is a version of the SB Audigy 2 NX firmware lacking - the audio fields in the endpoint descriptors */ - if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC || - (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bSynchAddress != 0 && - !implicit_fb)) { - snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", - dev->devnum, fmt->iface, fmt->altsetting, - get_endpoint(alts, 1)->bmAttributes, - get_endpoint(alts, 1)->bLength, - get_endpoint(alts, 1)->bSynchAddress); - return -EINVAL; - } - ep = get_endpoint(alts, 1)->bEndpointAddress; - if (!implicit_fb && - get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { - snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", - dev->devnum, fmt->iface, fmt->altsetting, - is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); - return -EINVAL; - } - - implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK) - == USB_ENDPOINT_USAGE_IMPLICIT_FB; + if (altsd->bNumEndpoints < 2) + return 0; - subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, - alts, ep, !subs->direction, - implicit_fb ? - SND_USB_ENDPOINT_TYPE_DATA : - SND_USB_ENDPOINT_TYPE_SYNC); - if (!subs->sync_endpoint) - return -EINVAL; + if ((is_playback && attr != USB_ENDPOINT_SYNC_ASYNC) || + (!is_playback && attr != USB_ENDPOINT_SYNC_ADAPTIVE)) + return 0; - subs->data_endpoint->sync_master = subs->sync_endpoint; + /* check sync-pipe endpoint */ + /* ... and check descriptor size before accessing bSynchAddress + because there is a version of the SB Audigy 2 NX firmware lacking + the audio fields in the endpoint descriptors */ + if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC || + (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + get_endpoint(alts, 1)->bSynchAddress != 0 && + !implicit_fb)) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + get_endpoint(alts, 1)->bmAttributes, + get_endpoint(alts, 1)->bLength, + get_endpoint(alts, 1)->bSynchAddress); + return -EINVAL; + } + ep = get_endpoint(alts, 1)->bEndpointAddress; + if (!implicit_fb && + get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); + return -EINVAL; } + implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK) + == USB_ENDPOINT_USAGE_IMPLICIT_FB; + + subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, + alts, ep, !subs->direction, + implicit_fb ? + SND_USB_ENDPOINT_TYPE_DATA : + SND_USB_ENDPOINT_TYPE_SYNC); + if (!subs->sync_endpoint) + return -EINVAL; + + subs->data_endpoint->sync_master = subs->sync_endpoint; + return 0; } -- cgit v0.10.2 From 95fec88332dbbe4344ffc1b564480402a89ee805 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sat, 3 Aug 2013 10:50:20 +0200 Subject: ALSA: usb-audio: do not initialize and check implicit_fb Since implicit_fb is not changed, !implicit_fb will always be true - it is set only after these checks. Similarly, there's also no need to set it at the top of the function. Change the type of implicit_fb to bool (more appropriate). Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c31dbdc..bb2e0f5 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -405,7 +405,7 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, { int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; unsigned int ep, attr; - int implicit_fb = 0; + bool implicit_fb; int err; /* we need a sync pipe in async OUT or adaptive IN mode */ @@ -432,8 +432,7 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, the audio fields in the endpoint descriptors */ if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC || (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && - get_endpoint(alts, 1)->bSynchAddress != 0 && - !implicit_fb)) { + get_endpoint(alts, 1)->bSynchAddress != 0)) { snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", dev->devnum, fmt->iface, fmt->altsetting, get_endpoint(alts, 1)->bmAttributes, @@ -442,8 +441,7 @@ static int set_sync_endpoint(struct snd_usb_substream *subs, return -EINVAL; } ep = get_endpoint(alts, 1)->bEndpointAddress; - if (!implicit_fb && - get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", -- cgit v0.10.2 From 914273c714845e2f3363e962f6dff59626a79fa3 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sat, 3 Aug 2013 10:50:21 +0200 Subject: ALSA: usb-audio: remove is_playback from implicit feedback quirks An implicit feedback endpoint can only be a capture source. The consumer (sink) of the implicit feedback endpoint is therefore limited to playback EPs. Check if the target endpoint is a playback first and remove redundant checks. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index bb2e0f5..af30e08 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -334,41 +334,39 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, { struct usb_host_interface *alts; struct usb_interface *iface; - int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; int implicit_fb = 0; unsigned int ep; + /* Implicit feedback sync EPs consumers are always playback EPs */ + if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK) + return 0; + switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ - if (is_playback) { - implicit_fb = 1; - ep = 0x81; - iface = usb_ifnum_to_if(dev, 3); + implicit_fb = 1; + ep = 0x81; + iface = usb_ifnum_to_if(dev, 3); - if (!iface || iface->num_altsetting == 0) - return -EINVAL; + if (!iface || iface->num_altsetting == 0) + return -EINVAL; - alts = &iface->altsetting[1]; - goto add_sync_ep; - } + alts = &iface->altsetting[1]; + goto add_sync_ep; break; case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ case USB_ID(0x0763, 0x2081): - if (is_playback) { - implicit_fb = 1; - ep = 0x81; - iface = usb_ifnum_to_if(dev, 2); + implicit_fb = 1; + ep = 0x81; + iface = usb_ifnum_to_if(dev, 2); - if (!iface || iface->num_altsetting == 0) - return -EINVAL; + if (!iface || iface->num_altsetting == 0) + return -EINVAL; - alts = &iface->altsetting[1]; - goto add_sync_ep; - } + alts = &iface->altsetting[1]; + goto add_sync_ep; } - if (is_playback && - attr == USB_ENDPOINT_SYNC_ASYNC && + if (attr == USB_ENDPOINT_SYNC_ASYNC && altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC && altsd->bInterfaceProtocol == 2 && altsd->bNumEndpoints == 1 && -- cgit v0.10.2 From 88abb8eff494d0be7819e744e74d62d5bc852905 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sat, 3 Aug 2013 10:51:14 +0200 Subject: ALSA: usb-audio: remove implicit_fb from quirk Since the quirks all apply to implicit feedback (the source endpoint is always a data endpoint), there's no need to set and check a flag for it. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index af30e08..b375d58 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -334,7 +334,6 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, { struct usb_host_interface *alts; struct usb_interface *iface; - int implicit_fb = 0; unsigned int ep; /* Implicit feedback sync EPs consumers are always playback EPs */ @@ -344,7 +343,6 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, switch (subs->stream->chip->usb_id) { case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */ case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */ - implicit_fb = 1; ep = 0x81; iface = usb_ifnum_to_if(dev, 3); @@ -356,7 +354,6 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, break; case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */ case USB_ID(0x0763, 0x2081): - implicit_fb = 1; ep = 0x81; iface = usb_ifnum_to_if(dev, 2); @@ -374,7 +371,6 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1, altsd->bAlternateSetting, &alts, &ep) >= 0) { - implicit_fb = 1; goto add_sync_ep; } @@ -384,9 +380,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, add_sync_ep: subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip, alts, ep, !subs->direction, - implicit_fb ? - SND_USB_ENDPOINT_TYPE_DATA : - SND_USB_ENDPOINT_TYPE_SYNC); + SND_USB_ENDPOINT_TYPE_DATA); if (!subs->sync_endpoint) return -EINVAL; -- cgit v0.10.2 From e7e58df8ef3c9edb09a240084b4e0523c12bcb71 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Sat, 3 Aug 2013 10:51:15 +0200 Subject: ALSA: usb-audio: WARN_ON when alts is passed as NULL Prevent NULL dereference in snd_usb_add_endpoints(), when alts is passed as NULL. In this case, WARN (since this is a non-fatal bug) and return NULL ep. Call sites treat a NULL return value as an error. Signed-off-by: Eldad Zack Signed-off-by: Takashi Iwai diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7a444b5..92ea945 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -418,6 +418,9 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep; int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK; + if (WARN_ON(!alts)) + return NULL; + mutex_lock(&chip->mutex); list_for_each_entry(ep, &chip->ep_list, list) { -- cgit v0.10.2 From 663819fb7d7e21f45431db1a2c0180bf2388ed2f Mon Sep 17 00:00:00 2001 From: Andy Shevchenko Date: Wed, 7 Aug 2013 17:55:14 +0300 Subject: ALSA: don't push static constants on stack for %*ph There is no need to pass constants via stack. The width may be explicitly specified in the format. Signed-off-by: Andy Shevchenko Signed-off-by: Takashi Iwai diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 9942691..afef0d7 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus) for (i = 0; i < 8; ++i) iwave[i] = snd_gf1_peek(gus, bank_pos + i); #ifdef CONFIG_SND_DEBUG_ROM - printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos, - 8, iwave); + printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave); #endif if (strncmp(iwave, "INTRWAVE", 8)) continue; /* first check */ diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index b9defcd..780bf3f 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -346,10 +346,10 @@ static int usb6fire_fw_check(u8 *version) if (!memcmp(version, known_fw_versions + i, 2)) return 0; - snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. " + snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. " "please reconnect to power. if this failure " "still happens, check your firmware installation.", - 4, version); + version); return -EINVAL; } -- cgit v0.10.2 From f5cb0be917ff30ef1df0aab5eb4cb269713c05e3 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 9 Aug 2013 12:57:06 +0300 Subject: sound: oss/dmabuf: remove an unneeded temporary variable We don't actually use the "go" variable so it can be removed. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index a59c888..461d94c 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -557,7 +557,6 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock) unsigned long flags; int err = 0, n = 0; struct dma_buffparms *dmap = adev->dmap_in; - int go; if (!(adev->open_mode & OPEN_READ)) return -EIO; @@ -584,7 +583,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock) spin_unlock_irqrestore(&dmap->lock,flags); return -EAGAIN; } - if ((go = adev->go)) + if (adev->go) timeout = dmabuf_timeout(dmap); spin_unlock_irqrestore(&dmap->lock,flags); -- cgit v0.10.2 From 33bbe1499cb3eebb4d5f66d56ed8026d0bc56d63 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 9 Aug 2013 18:10:50 +0100 Subject: ALSA: Add MAINTAINERS entry for compressed audio API Help ensure that Vinod gets included in review of compressed audio patches by adding a MAINTAINERS entry for it. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai diff --git a/MAINTAINERS b/MAINTAINERS index bf61e04..b4d1de9 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7655,6 +7655,15 @@ F: include/sound/ F: include/uapi/sound/ F: sound/ +SOUND - COMPRESSED AUDIO +M: Vinod Koul +L: alsa-devel@alsa-project.org (moderated for non-subscribers) +T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git +S: Supported +F: include/sound/compress_driver.h +F: sound/core/compress_offload.c +F: sound/soc/soc-compress.c + SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood M: Mark Brown -- cgit v0.10.2 From f672f31ab58c1e7e96acb4ea54eebb8bb59a2667 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 12 Aug 2013 11:15:27 +0530 Subject: ALSA: compress: update the MAINTAINER entry add missing uapi/ headers and Documentation files Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai diff --git a/MAINTAINERS b/MAINTAINERS index b4d1de9..1c2bba3 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7660,7 +7660,9 @@ M: Vinod Koul L: alsa-devel@alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Supported +F: Documentation/sound/alsa/compress_offload.txt F: include/sound/compress_driver.h +F: include/uapi/sound/compress_* F: sound/core/compress_offload.c F: sound/soc/soc-compress.c -- cgit v0.10.2 From bc2eee29fc8224ffad495d0c68ead0ce603309e3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Aug 2013 15:05:03 +0200 Subject: ALSA: hda - Allow auto_mute_via_amp on bind mute controls The auto-mute using the amp currently works only for a single amp on a pin. Make it working also with HDA_CTL_BIND_MUTE type, too. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index f6c0344..6ed2209 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -816,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx) static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); enum { HDA_CTL_WIDGET_VOL, @@ -833,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = { .put = hda_gen_mixer_mute_put, /* replaced */ .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), }, - HDA_BIND_MUTE(NULL, 0, 0, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_bind_switch_get, + .put = hda_gen_bind_mute_put, /* replaced */ + .private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0), + }, }; /* add dynamic controls from template */ @@ -940,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx, } /* playback mute control with the software mute bit check */ -static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_gen_spec *spec = codec->spec; @@ -952,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] &= enabled; ucontrol->value.integer.value[1] &= enabled; } +} +static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } +static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + sync_auto_mute_bits(kcontrol, ucontrol); + return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol); +} + /* any ctl assigned to the path with the given index? */ static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type) { -- cgit v0.10.2 From e80c60f3cbe76fa95029abc53b1a29172b51b96a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 12 Aug 2013 14:44:59 +0200 Subject: ALSA: hda - Mute the right widget in auto_mute_via_amp mode The current generic parser code assumes that always a pin widget controls the mute for an output blindly although it might be a different widget in the middle. Instead of the fixed assumption, check each parsed path and just pick up the right widget that has been already defined as a mute control. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 6ed2209..fd1965c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3768,7 +3768,7 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins) /* standard HP/line-out auto-mute helper */ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, - bool mute) + int *paths, bool mute) { struct hda_gen_spec *spec = codec->spec; int i; @@ -3780,10 +3780,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, break; if (spec->auto_mute_via_amp) { + struct nid_path *path; + hda_nid_t mute_nid; + + path = snd_hda_get_path_from_idx(codec, paths[i]); + if (!path) + continue; + mute_nid = get_amp_nid_(path->ctls[NID_PATH_MUTE_CTL]); + if (!mute_nid) + continue; if (mute) - spec->mute_bits |= (1ULL << nid); + spec->mute_bits |= (1ULL << mute_nid); else - spec->mute_bits &= ~(1ULL << nid); + spec->mute_bits &= ~(1ULL << mute_nid); set_pin_eapd(codec, nid, !mute); continue; } @@ -3814,14 +3823,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, void snd_hda_gen_update_outputs(struct hda_codec *codec) { struct hda_gen_spec *spec = codec->spec; + int *paths; int on; /* Control HP pins/amps depending on master_mute state; * in general, HP pins/amps control should be enabled in all cases, * but currently set only for master_mute, just to be safe */ + if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT) + paths = spec->out_paths; + else + paths = spec->hp_paths; do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), - spec->autocfg.hp_pins, spec->master_mute); + spec->autocfg.hp_pins, paths, spec->master_mute); if (!spec->automute_speaker) on = 0; @@ -3829,8 +3843,12 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec) on = spec->hp_jack_present | spec->line_jack_present; on |= spec->master_mute; spec->speaker_muted = on; + if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT) + paths = spec->out_paths; + else + paths = spec->speaker_paths; do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins), - spec->autocfg.speaker_pins, on); + spec->autocfg.speaker_pins, paths, on); /* toggle line-out mutes if needed, too */ /* if LO is a copy of either HP or Speaker, don't need to handle it */ @@ -3843,8 +3861,9 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec) on = spec->hp_jack_present; on |= spec->master_mute; spec->line_out_muted = on; + paths = spec->out_paths; do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), - spec->autocfg.line_out_pins, on); + spec->autocfg.line_out_pins, paths, on); } EXPORT_SYMBOL_HDA(snd_hda_gen_update_outputs); -- cgit v0.10.2 From b892ca1c9fe71e829e7b9ed79b8398649de259d7 Mon Sep 17 00:00:00 2001 From: Knut Petersen Date: Tue, 13 Aug 2013 21:18:12 +0200 Subject: ALSA: rme96: Add pcm stream synchronization The hardware does support synchronized start/pause/stop of pcm streams, so there is no reason not to add that feature after more than ten years. Some minor coding style / white space fixes in the surroundings of the changes. Signed-off-by: Knut Petersen Signed-off-by: Takashi Iwai diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 2a8ad9d..4e9a556 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -198,6 +198,31 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard."); #define RME96_AD1852_VOL_BITS 14 #define RME96_AD1855_VOL_BITS 10 +/* Defines for snd_rme96_trigger */ +#define RME96_TB_START_PLAYBACK 1 +#define RME96_TB_START_CAPTURE 2 +#define RME96_TB_STOP_PLAYBACK 4 +#define RME96_TB_STOP_CAPTURE 8 +#define RME96_TB_RESET_PLAYPOS 16 +#define RME96_TB_RESET_CAPTUREPOS 32 +#define RME96_TB_CLEAR_PLAYBACK_IRQ 64 +#define RME96_TB_CLEAR_CAPTURE_IRQ 128 +#define RME96_RESUME_PLAYBACK (RME96_TB_START_PLAYBACK) +#define RME96_RESUME_CAPTURE (RME96_TB_START_CAPTURE) +#define RME96_RESUME_BOTH (RME96_RESUME_PLAYBACK \ + | RME96_RESUME_CAPTURE) +#define RME96_START_PLAYBACK (RME96_TB_START_PLAYBACK \ + | RME96_TB_RESET_PLAYPOS) +#define RME96_START_CAPTURE (RME96_TB_START_CAPTURE \ + | RME96_TB_RESET_CAPTUREPOS) +#define RME96_START_BOTH (RME96_START_PLAYBACK \ + | RME96_START_CAPTURE) +#define RME96_STOP_PLAYBACK (RME96_TB_STOP_PLAYBACK \ + | RME96_TB_CLEAR_PLAYBACK_IRQ) +#define RME96_STOP_CAPTURE (RME96_TB_STOP_CAPTURE \ + | RME96_TB_CLEAR_CAPTURE_IRQ) +#define RME96_STOP_BOTH (RME96_STOP_PLAYBACK \ + | RME96_STOP_CAPTURE) struct rme96 { spinlock_t lock; @@ -344,6 +369,7 @@ static struct snd_pcm_hardware snd_rme96_playback_spdif_info = { .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -373,6 +399,7 @@ static struct snd_pcm_hardware snd_rme96_capture_spdif_info = { .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -402,6 +429,7 @@ static struct snd_pcm_hardware snd_rme96_playback_adat_info = { .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -427,6 +455,7 @@ static struct snd_pcm_hardware snd_rme96_capture_adat_info = { .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -1045,54 +1074,35 @@ snd_rme96_capture_hw_params(struct snd_pcm_substream *substream, } static void -snd_rme96_playback_start(struct rme96 *rme96, - int from_pause) +snd_rme96_trigger(struct rme96 *rme96, + int op) { - if (!from_pause) { + if (op & RME96_TB_RESET_PLAYPOS) writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS); - } - - rme96->wcreg |= RME96_WCR_START; - writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); -} - -static void -snd_rme96_capture_start(struct rme96 *rme96, - int from_pause) -{ - if (!from_pause) { + if (op & RME96_TB_RESET_CAPTUREPOS) writel(0, rme96->iobase + RME96_IO_RESET_REC_POS); - } - - rme96->wcreg |= RME96_WCR_START_2; + if (op & RME96_TB_CLEAR_PLAYBACK_IRQ) { + rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); + if (rme96->rcreg & RME96_RCR_IRQ) + writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ); + } + if (op & RME96_TB_CLEAR_CAPTURE_IRQ) { + rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); + if (rme96->rcreg & RME96_RCR_IRQ_2) + writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ); + } + if (op & RME96_TB_START_PLAYBACK) + rme96->wcreg |= RME96_WCR_START; + if (op & RME96_TB_STOP_PLAYBACK) + rme96->wcreg &= ~RME96_WCR_START; + if (op & RME96_TB_START_CAPTURE) + rme96->wcreg |= RME96_WCR_START_2; + if (op & RME96_TB_STOP_CAPTURE) + rme96->wcreg &= ~RME96_WCR_START_2; writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); } -static void -snd_rme96_playback_stop(struct rme96 *rme96) -{ - /* - * Check if there is an unconfirmed IRQ, if so confirm it, or else - * the hardware will not stop generating interrupts - */ - rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); - if (rme96->rcreg & RME96_RCR_IRQ) { - writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ); - } - rme96->wcreg &= ~RME96_WCR_START; - writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); -} -static void -snd_rme96_capture_stop(struct rme96 *rme96) -{ - rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER); - if (rme96->rcreg & RME96_RCR_IRQ_2) { - writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ); - } - rme96->wcreg &= ~RME96_WCR_START_2; - writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER); -} static irqreturn_t snd_rme96_interrupt(int irq, @@ -1155,6 +1165,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_set_sync(substream); spin_lock_irq(&rme96->lock); if (rme96->playback_substream != NULL) { spin_unlock_irq(&rme96->lock); @@ -1191,6 +1202,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_set_sync(substream); runtime->hw = snd_rme96_capture_spdif_info; if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG && (rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0) @@ -1222,6 +1234,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_set_sync(substream); spin_lock_irq(&rme96->lock); if (rme96->playback_substream != NULL) { spin_unlock_irq(&rme96->lock); @@ -1253,6 +1266,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream) struct rme96 *rme96 = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_set_sync(substream); runtime->hw = snd_rme96_capture_adat_info; if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) { /* makes no sense to use analog input. Note that analog @@ -1288,7 +1302,7 @@ snd_rme96_playback_close(struct snd_pcm_substream *substream) spin_lock_irq(&rme96->lock); if (RME96_ISPLAYING(rme96)) { - snd_rme96_playback_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK); } rme96->playback_substream = NULL; rme96->playback_periodsize = 0; @@ -1309,7 +1323,7 @@ snd_rme96_capture_close(struct snd_pcm_substream *substream) spin_lock_irq(&rme96->lock); if (RME96_ISRECORDING(rme96)) { - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_CAPTURE); } rme96->capture_substream = NULL; rme96->capture_periodsize = 0; @@ -1324,7 +1338,7 @@ snd_rme96_playback_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&rme96->lock); if (RME96_ISPLAYING(rme96)) { - snd_rme96_playback_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK); } writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS); spin_unlock_irq(&rme96->lock); @@ -1338,7 +1352,7 @@ snd_rme96_capture_prepare(struct snd_pcm_substream *substream) spin_lock_irq(&rme96->lock); if (RME96_ISRECORDING(rme96)) { - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_CAPTURE); } writel(0, rme96->iobase + RME96_IO_RESET_REC_POS); spin_unlock_irq(&rme96->lock); @@ -1350,41 +1364,53 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct rme96 *rme96 = snd_pcm_substream_chip(substream); + struct snd_pcm_substream *s; + bool sync; + + snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) == rme96) + snd_pcm_trigger_done(s, substream); + } + + sync = (rme96->playback_substream && rme96->capture_substream) && + (rme96->playback_substream->group == + rme96->capture_substream->group); switch (cmd) { case SNDRV_PCM_TRIGGER_START: if (!RME96_ISPLAYING(rme96)) { - if (substream != rme96->playback_substream) { + if (substream != rme96->playback_substream) return -EBUSY; - } - snd_rme96_playback_start(rme96, 0); + snd_rme96_trigger(rme96, sync ? RME96_START_BOTH + : RME96_START_PLAYBACK); } break; case SNDRV_PCM_TRIGGER_STOP: if (RME96_ISPLAYING(rme96)) { - if (substream != rme96->playback_substream) { + if (substream != rme96->playback_substream) return -EBUSY; - } - snd_rme96_playback_stop(rme96); + snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH + : RME96_STOP_PLAYBACK); } break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (RME96_ISPLAYING(rme96)) { - snd_rme96_playback_stop(rme96); - } + if (RME96_ISPLAYING(rme96)) + snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH + : RME96_STOP_PLAYBACK); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!RME96_ISPLAYING(rme96)) { - snd_rme96_playback_start(rme96, 1); - } + if (!RME96_ISPLAYING(rme96)) + snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH + : RME96_RESUME_PLAYBACK); break; - + default: return -EINVAL; } + return 0; } @@ -1393,38 +1419,49 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream, int cmd) { struct rme96 *rme96 = snd_pcm_substream_chip(substream); + struct snd_pcm_substream *s; + bool sync; + + snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) == rme96) + snd_pcm_trigger_done(s, substream); + } + + sync = (rme96->playback_substream && rme96->capture_substream) && + (rme96->playback_substream->group == + rme96->capture_substream->group); switch (cmd) { case SNDRV_PCM_TRIGGER_START: if (!RME96_ISRECORDING(rme96)) { - if (substream != rme96->capture_substream) { + if (substream != rme96->capture_substream) return -EBUSY; - } - snd_rme96_capture_start(rme96, 0); + snd_rme96_trigger(rme96, sync ? RME96_START_BOTH + : RME96_START_CAPTURE); } break; case SNDRV_PCM_TRIGGER_STOP: if (RME96_ISRECORDING(rme96)) { - if (substream != rme96->capture_substream) { + if (substream != rme96->capture_substream) return -EBUSY; - } - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH + : RME96_STOP_CAPTURE); } break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (RME96_ISRECORDING(rme96)) { - snd_rme96_capture_stop(rme96); - } + if (RME96_ISRECORDING(rme96)) + snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH + : RME96_STOP_CAPTURE); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!RME96_ISRECORDING(rme96)) { - snd_rme96_capture_start(rme96, 1); - } + if (!RME96_ISRECORDING(rme96)) + snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH + : RME96_RESUME_CAPTURE); break; - + default: return -EINVAL; } @@ -1505,8 +1542,7 @@ snd_rme96_free(void *private_data) return; } if (rme96->irq >= 0) { - snd_rme96_playback_stop(rme96); - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_BOTH); rme96->areg &= ~RME96_AR_DAC_EN; writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); free_irq(rme96->irq, (void *)rme96); @@ -1606,8 +1642,7 @@ snd_rme96_create(struct rme96 *rme96) rme96->capture_periodsize = 0; /* make sure playback/capture is stopped, if by some reason active */ - snd_rme96_playback_stop(rme96); - snd_rme96_capture_stop(rme96); + snd_rme96_trigger(rme96, RME96_STOP_BOTH); /* set default values in registers */ rme96->wcreg = -- cgit v0.10.2 From a4a9e082671d2f1e9d3b49a3692313087b036aff Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 16 Aug 2013 14:09:01 +0200 Subject: ALSA: hda - Fix internal mic boost on three Thinkpad machines The internal mic boost is so noisy on boosts 2 and 3 so they are unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213055 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6ac4810..333d1a6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3694,6 +3694,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- cgit v0.10.2 From aaedfb4761697e6fe24a7443e8d288636ccc69ef Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Fri, 16 Aug 2013 14:09:02 +0200 Subject: ALSA: hda - Fix the order of a quirk table (janitorial) This just cleans up the table, no functional changes. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 333d1a6..96dcb68 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3622,6 +3622,11 @@ static const struct hda_fixup alc269_fixups[] = { static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), + SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), + SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), + SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), + SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), @@ -3677,11 +3682,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), - SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), - SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), - SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), - SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), - SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), @@ -3692,8 +3692,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- cgit v0.10.2 From cd5302c0d4b79bef7660bb4be300d169e38f39c3 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 19 Aug 2013 12:22:33 +0200 Subject: ALSA: hda - Limit internal mic boost for a few more Thinkpad machines The higher mic boosts (on internal mic) are so noisy they're unusable in practice. BugLink: https://bugs.launchpad.net/bugs/1213820 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 96dcb68..7d00639 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3694,9 +3694,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ -- cgit v0.10.2 From c841ad2a9b86c7317dc7e4fe4e03bc56a6c0d6e8 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 19 Aug 2013 13:32:30 +0200 Subject: ALSA: hda - Try to allow haswell HDMI audio even without powerwell If compiled without CONFIG_SND_HDA_I915, the audio driver cannot request power well. However, if the power well is on for other reasons, maybe audio can still work. Therefore, do not skip the card completely if compiled without CONFIG_SND_HDA_I915. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7f9e406..c6c9829 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3855,11 +3855,13 @@ static int azx_probe_continue(struct azx *chip) /* Request power well for Haswell HDA controller and codec */ if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { +#ifdef CONFIG_SND_HDA_I915 err = hda_i915_init(); if (err < 0) { snd_printk(KERN_ERR SFX "Error request power-well from i915\n"); goto out_free; } +#endif hda_display_power(true); } -- cgit v0.10.2 From bcbb15530ebe9737622cb7779b35c61f48b49734 Mon Sep 17 00:00:00 2001 From: Tim Gardner Date: Fri, 16 Aug 2013 11:18:59 -0600 Subject: ALSA: pcm: Add snd_printd_ratelimit() Direct calls to printk_limit() will emit log noise even when CONFIG_SND_DEBUG is not defined. Add a wrapper macro around printk_limit() that is conditionally defined by CONFIG_SND_DEBUG. Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Pierre-Louis Bossart Cc: Lars-Peter Clausen Cc: Yacine Belkadi Signed-off-by: Tim Gardner Signed-off-by: Takashi Iwai diff --git a/include/sound/core.h b/include/sound/core.h index c586617..2a14f1f 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -27,6 +27,7 @@ #include /* struct rw_semaphore */ #include /* pm_message_t */ #include +#include /* number of supported soundcards */ #ifdef CONFIG_SND_DYNAMIC_MINORS @@ -376,6 +377,11 @@ void __snd_printk(unsigned int level, const char *file, int line, #define snd_BUG() WARN(1, "BUG?\n") /** + * Suppress high rates of output when CONFIG_SND_DEBUG is enabled. + */ +#define snd_printd_ratelimit() printk_ratelimit() + +/** * snd_BUG_ON - debugging check macro * @cond: condition to evaluate * @@ -398,6 +404,8 @@ static inline void _snd_printd(int level, const char *format, ...) {} unlikely(__ret_warn_on); \ }) +static inline bool snd_printd_ratelimit(void) { return false; } + #endif /* CONFIG_SND_DEBUG */ #ifdef CONFIG_SND_DEBUG_VERBOSE -- cgit v0.10.2 From 74d779ab7c9f9024cfead259206e0e0b20ee37e4 Mon Sep 17 00:00:00 2001 From: Tim Gardner Date: Fri, 16 Aug 2013 11:19:00 -0600 Subject: ALSA: pcm: Use snd_printd_ratelimit() The use of snd_printd_ratelimit() supresses superfluous output from printk_ratelimit() when CONFIG_SND_DEBUG is not defined. For example, [ 43.753692] snd_pcm_update_hw_ptr0: 26 callbacks suppressed [ 48.822131] snd_pcm_update_hw_ptr0: 25 callbacks suppressed [ 53.894953] snd_pcm_update_hw_ptr0: 25 callbacks suppressed [ 58.997761] snd_pcm_update_hw_ptr0: 25 callbacks suppressed [ 64.100952] snd_pcm_update_hw_ptr0: 25 callbacks suppressed fills the log even when no debug output is actually produced. Cc: Jaroslav Kysela Cc: Takashi Iwai Cc: Pierre-Louis Bossart Cc: Lars-Peter Clausen Cc: Yacine Belkadi Signed-off-by: Tim Gardner Signed-off-by: Takashi Iwai diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 82bb029..6e03b46 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -184,7 +184,7 @@ static void xrun(struct snd_pcm_substream *substream) do { \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ xrun_log_show(substream); \ - if (printk_ratelimit()) { \ + if (snd_printd_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ dump_stack_on_xrun(substream); \ @@ -342,7 +342,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, return -EPIPE; } if (pos >= runtime->buffer_size) { - if (printk_ratelimit()) { + if (snd_printd_ratelimit()) { char name[16]; snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); -- cgit v0.10.2 From 17d2f00836cce9b1a24e65670ad78dbab275777b Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Mon, 19 Aug 2013 17:20:30 +0200 Subject: ALSA: hdspm - Fix default value in SNDRV_HDSPM_IOCTL_GET_LTC Use enum hdspm_ltc_format's fps_30 (corresponds to 4) instead of 30, Other case branches return 1, 2 or 3 respectively, so 30 obviously is wrong. Since SNDRV_HDSPM_IOCTL_GET_LTC had never been working due to a copy&paste error in hdspm.h, this change doesn't break userspace. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ec6335e..e4d76a6 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -6311,7 +6311,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, ltc.format = fps_2997; break; default: - ltc.format = 30; + ltc.format = fps_30; break; } if (i & HDSPM_TCO1_set_drop_frame_flag) { -- cgit v0.10.2 From 1568b8802227f4e7b0ad79a49cd35d4e285570f2 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Mon, 19 Aug 2013 17:20:31 +0200 Subject: ALSA: hdspm - Use enums in hdspm_tco_ltc_frames() This patch doesn't change functionality, it only improves readability and fixes a copy&paste error in a comment. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index e4d76a6..3cde55b 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -3173,19 +3173,19 @@ static int hdspm_tco_ltc_frames(struct hdspm *hdspm) HDSPM_TCO1_LTC_Format_MSB)) { case 0: /* 24 fps */ - ret = 1; + ret = fps_24; break; case HDSPM_TCO1_LTC_Format_LSB: /* 25 fps */ - ret = 2; + ret = fps_25; break; case HDSPM_TCO1_LTC_Format_MSB: - /* 25 fps */ - ret = 3; + /* 29.97 fps */ + ret = fps_2997; break; default: /* 30 fps */ - ret = 4; + ret = fps_30; break; } } -- cgit v0.10.2 From b43dd416be21bc8ad60984e13def032f01aaaa18 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Mon, 19 Aug 2013 17:20:32 +0200 Subject: ALSA: hdspm - Fix SNDRV_HDSPM_IOCTL_GET_LTC Use struct hdspm_ltc to query the LTC, using a mixer struct is just plain wrong. Due to the wrong struct, this ioctl was never working, so we're free to fix it without breaking userspace compatibility. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h index 1f59ea2..d956c35 100644 --- a/include/uapi/sound/hdspm.h +++ b/include/uapi/sound/hdspm.h @@ -111,7 +111,7 @@ struct hdspm_ltc { enum hdspm_ltc_input_format input_format; }; -#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl) +#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_ltc) /** * The status data reflects the device's current state -- cgit v0.10.2 From 528ba522e18b95d25adc62367f04290776c390e5 Mon Sep 17 00:00:00 2001 From: Knut Petersen Date: Wed, 21 Aug 2013 09:18:54 +0200 Subject: ALSA: rme96: Add PM support v3 Without proper power management handling, the first use of a Digi96/8 anytime after a suspend / resume cycle will start playback with distortions. v3: Abort if vmalloc() of suspend buffers fail, but do not leak memory in that case. [fixed wrong memory leak fix again -- tiwai] Signed-off-by: Knut Petersen Signed-off-by: Takashi Iwai diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 4e9a556..0506530 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -239,6 +239,13 @@ struct rme96 { u8 rev; /* card revision number */ +#ifdef CONFIG_PM + u32 playback_pointer; + u32 capture_pointer; + void *playback_suspend_buffer; + void *capture_suspend_buffer; +#endif + struct snd_pcm_substream *playback_substream; struct snd_pcm_substream *capture_substream; @@ -370,6 +377,7 @@ static struct snd_pcm_hardware snd_rme96_playback_spdif_info = .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -400,6 +408,7 @@ static struct snd_pcm_hardware snd_rme96_capture_spdif_info = .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -430,6 +439,7 @@ static struct snd_pcm_hardware snd_rme96_playback_adat_info = .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -456,6 +466,7 @@ static struct snd_pcm_hardware snd_rme96_capture_adat_info = .info = (SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE), .formats = (SNDRV_PCM_FMTBIT_S16_LE | @@ -1386,6 +1397,7 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream, } break; + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: if (RME96_ISPLAYING(rme96)) { if (substream != rme96->playback_substream) @@ -1401,6 +1413,7 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream, : RME96_STOP_PLAYBACK); break; + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!RME96_ISPLAYING(rme96)) snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH @@ -1441,6 +1454,7 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream, } break; + case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: if (RME96_ISRECORDING(rme96)) { if (substream != rme96->capture_substream) @@ -1456,6 +1470,7 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream, : RME96_STOP_CAPTURE); break; + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!RME96_ISRECORDING(rme96)) snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH @@ -1556,6 +1571,10 @@ snd_rme96_free(void *private_data) pci_release_regions(rme96->pci); rme96->port = 0; } +#ifdef CONFIG_PM + vfree(rme96->playback_suspend_buffer); + vfree(rme96->capture_suspend_buffer); +#endif pci_disable_device(rme96->pci); } @@ -2354,6 +2373,83 @@ snd_rme96_create_switches(struct snd_card *card, * Card initialisation */ +#ifdef CONFIG_PM + +static int +snd_rme96_suspend(struct pci_dev *pci, + pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct rme96 *rme96 = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend(rme96->playback_substream); + snd_pcm_suspend(rme96->capture_substream); + + /* save capture & playback pointers */ + rme96->playback_pointer = readl(rme96->iobase + RME96_IO_GET_PLAY_POS) + & RME96_RCR_AUDIO_ADDR_MASK; + rme96->capture_pointer = readl(rme96->iobase + RME96_IO_GET_REC_POS) + & RME96_RCR_AUDIO_ADDR_MASK; + + /* save playback and capture buffers */ + memcpy_fromio(rme96->playback_suspend_buffer, + rme96->iobase + RME96_IO_PLAY_BUFFER, RME96_BUFFER_SIZE); + memcpy_fromio(rme96->capture_suspend_buffer, + rme96->iobase + RME96_IO_REC_BUFFER, RME96_BUFFER_SIZE); + + /* disable the DAC */ + rme96->areg &= ~RME96_AR_DAC_EN; + writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); + + pci_disable_device(pci); + pci_save_state(pci); + + return 0; +} + +static int +snd_rme96_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct rme96 *rme96 = card->private_data; + + pci_restore_state(pci); + pci_enable_device(pci); + + /* reset playback and record buffer pointers */ + writel(0, rme96->iobase + RME96_IO_SET_PLAY_POS + + rme96->playback_pointer); + writel(0, rme96->iobase + RME96_IO_SET_REC_POS + + rme96->capture_pointer); + + /* restore playback and capture buffers */ + memcpy_toio(rme96->iobase + RME96_IO_PLAY_BUFFER, + rme96->playback_suspend_buffer, RME96_BUFFER_SIZE); + memcpy_toio(rme96->iobase + RME96_IO_REC_BUFFER, + rme96->capture_suspend_buffer, RME96_BUFFER_SIZE); + + /* reset the ADC */ + writel(rme96->areg | RME96_AR_PD2, + rme96->iobase + RME96_IO_ADDITIONAL_REG); + writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); + + /* reset and enable DAC, restore analog volume */ + snd_rme96_reset_dac(rme96); + rme96->areg |= RME96_AR_DAC_EN; + writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG); + if (RME96_HAS_ANALOG_OUT(rme96)) { + usleep_range(3000, 10000); + snd_rme96_apply_dac_volume(rme96); + } + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} + +#endif + static void snd_rme96_card_free(struct snd_card *card) { snd_rme96_free(card->private_data); @@ -2390,6 +2486,23 @@ snd_rme96_probe(struct pci_dev *pci, return err; } +#ifdef CONFIG_PM + rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); + if (!rme96->playback_suspend_buffer) { + snd_printk(KERN_ERR + "Failed to allocate playback suspend buffer!\n"); + snd_card_free(card); + return -ENOMEM; + } + rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE); + if (!rme96->capture_suspend_buffer) { + snd_printk(KERN_ERR + "Failed to allocate capture suspend buffer!\n"); + snd_card_free(card); + return -ENOMEM; + } +#endif + strcpy(card->driver, "Digi96"); switch (rme96->pci->device) { case PCI_DEVICE_ID_RME_DIGI96: @@ -2432,6 +2545,10 @@ static struct pci_driver rme96_driver = { .id_table = snd_rme96_ids, .probe = snd_rme96_probe, .remove = snd_rme96_remove, +#ifdef CONFIG_PM + .suspend = snd_rme96_suspend, + .resume = snd_rme96_resume, +#endif }; module_pci_driver(rme96_driver); -- cgit v0.10.2 From 2af02be71a8ae28ae4e3b82a2866b1aa1f43d8fb Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 22 Aug 2013 10:03:50 +0200 Subject: ALSA: hda - Fix ALC283 headphone pop-noise better Fixed ALC283 D3 to D0 and D0 to D3 Headphone pop noise. The previous fix [c5177c86: ALSA: hda - Fix the noise after suspend on ALC283 codec] doesn't work sufficiently for some laptops. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7d00639..4bdccd1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2527,6 +2527,7 @@ enum { ALC269_TYPE_ALC269VD, ALC269_TYPE_ALC280, ALC269_TYPE_ALC282, + ALC269_TYPE_ALC283, ALC269_TYPE_ALC284, ALC269_TYPE_ALC286, }; @@ -2552,6 +2553,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec) case ALC269_TYPE_ALC269VB: case ALC269_TYPE_ALC269VD: case ALC269_TYPE_ALC282: + case ALC269_TYPE_ALC283: case ALC269_TYPE_ALC286: ssids = alc269_ssids; break; @@ -2586,6 +2588,74 @@ static void alc269_shutup(struct hda_codec *codec) snd_hda_shutup_pins(codec); } +static void alc283_init(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + bool hp_pin_sense; + int val; + + if (!hp_pin) + return; + hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); + + /* Index 0x43 Direct Drive HP AMP LPM Control 1 */ + /* Headphone capless set to high power mode */ + alc_write_coef_idx(codec, 0x43, 0x9004); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + if (hp_pin_sense) + msleep(85); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + + if (hp_pin_sense) + msleep(85); + /* Index 0x46 Combo jack auto switch control 2 */ + /* 3k pull low control for Headset jack. */ + val = alc_read_coef_idx(codec, 0x46); + alc_write_coef_idx(codec, 0x46, val & ~(3 << 12)); + /* Headphone capless set to normal mode */ + alc_write_coef_idx(codec, 0x43, 0x9614); +} + +static void alc283_shutup(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; + bool hp_pin_sense; + int val; + + if (!hp_pin) { + alc269_shutup(codec); + return; + } + + hp_pin_sense = snd_hda_jack_detect(codec, hp_pin); + + alc_write_coef_idx(codec, 0x43, 0x9004); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + + if (hp_pin_sense) + msleep(85); + + snd_hda_codec_write(codec, hp_pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); + + val = alc_read_coef_idx(codec, 0x46); + alc_write_coef_idx(codec, 0x46, val | (3 << 12)); + + if (hp_pin_sense) + msleep(85); + snd_hda_shutup_pins(codec); + alc_write_coef_idx(codec, 0x43, 0x9614); +} + static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg, unsigned int val) { @@ -2715,12 +2785,6 @@ static int alc269_resume(struct hda_codec *codec) if (spec->has_alc5505_dsp) alc5505_dsp_resume(codec); - /* clear the power-save mode for ALC283 */ - if (codec->vendor_id == 0x10ec0283) { - alc_write_coef_idx(codec, 0x4, 0xaf01); - alc_write_coef_idx(codec, 0x6, 0x2104); - } - return 0; } #endif /* CONFIG_PM */ @@ -3815,30 +3879,6 @@ static void alc269_fill_coef(struct hda_codec *codec) alc_write_coef_idx(codec, 0x4, val | (1<<11)); } -/* don't clear mic pin; otherwise it results in noise in D3 */ -static void alc283_headset_shutup(struct hda_codec *codec) -{ - int i; - - if (codec->bus->shutdown) - return; - - for (i = 0; i < codec->init_pins.used; i++) { - struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); - /* use read here for syncing after issuing each verb */ - if (pin->nid != 0x19) - snd_hda_codec_read(codec, pin->nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } - - alc_write_coef_idx(codec, 0x4, 0x0f01); /* power save */ - alc_write_coef_idx(codec, 0x6, 0x2100); /* power save */ - snd_hda_codec_write(codec, 0x19, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_VREFHIZ); - codec->pins_shutup = 1; -} - /* */ static int patch_alc269(struct hda_codec *codec) @@ -3853,9 +3893,6 @@ static int patch_alc269(struct hda_codec *codec) spec = codec->spec; spec->gen.shared_mic_vref_pin = 0x18; - if (codec->vendor_id == 0x10ec0283) - spec->shutup = alc283_headset_shutup; - snd_hda_pick_fixup(codec, alc269_fixup_models, alc269_fixup_tbl, alc269_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); @@ -3897,11 +3934,15 @@ static int patch_alc269(struct hda_codec *codec) case 0x10ec0290: spec->codec_variant = ALC269_TYPE_ALC280; break; - case 0x10ec0233: case 0x10ec0282: - case 0x10ec0283: spec->codec_variant = ALC269_TYPE_ALC282; break; + case 0x10ec0233: + case 0x10ec0283: + spec->codec_variant = ALC269_TYPE_ALC283; + spec->shutup = alc283_shutup; + spec->init_hook = alc283_init; + break; case 0x10ec0284: case 0x10ec0292: spec->codec_variant = ALC269_TYPE_ALC284; -- cgit v0.10.2 From cd217a6395ae1b14cd70908e190f566b8bbd282f Mon Sep 17 00:00:00 2001 From: Kailang Yang Date: Thu, 22 Aug 2013 10:15:24 +0200 Subject: ALSA: hda - Add workarounds for pop-noise on Chromebook with ALC283 The headphone automute on this machine triggers annoying pop noises. It seems that only the first DAC can be used, the secondary DAC always results in this problem. This patch disables the secondary DAC with a few additional workarounds. Signed-off-by: Kailang Yang Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4bdccd1..134fbe8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3394,6 +3394,45 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, } } +static void alc283_hp_automute_hook(struct hda_codec *codec, + struct hda_jack_tbl *jack) +{ + struct alc_spec *spec = codec->spec; + int vref; + + msleep(200); + snd_hda_gen_hp_automute(codec, jack); + + vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0; + + msleep(600); + snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + vref); +} + +static void alc283_chromebook_caps(struct hda_codec *codec) +{ + snd_hda_override_wcaps(codec, 0x03, 0); +} + +static void alc283_fixup_chromebook(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + int val; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + alc283_chromebook_caps(codec); + spec->gen.hp_automute_hook = alc283_hp_automute_hook; + /* MIC2-VREF control */ + /* Set to manual mode */ + val = alc_read_coef_idx(codec, 0x06); + alc_write_coef_idx(codec, 0x06, val & ~0x000c); + break; + } +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3430,6 +3469,7 @@ enum { ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ORDISSIMO_EVE2, + ALC283_FIXUP_CHROME_BOOK, }; static const struct hda_fixup alc269_fixups[] = { @@ -3681,6 +3721,10 @@ static const struct hda_fixup alc269_fixups[] = { { } }, }, + [ALC283_FIXUP_CHROME_BOOK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc283_fixup_chromebook, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3728,6 +3772,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- cgit v0.10.2 From e58a244ff9ae264df1bf0fc8f09ecc135dbe3d0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 22 Aug 2013 12:02:31 +0200 Subject: ALSA: rme96: Check the return value of pci_enable_device() in resume callback MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fixing warning message: sound/pci/rme96.c: In function ‘snd_rme96_resume’: sound/pci/rme96.c:2418:19: warning: ignoring return value of ‘pci_enable_device’, declared with attribute warn_unused_result [-Wunused-result] Signed-off-by: Takashi Iwai diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 0506530..9d2a81f 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -2415,7 +2415,11 @@ snd_rme96_resume(struct pci_dev *pci) struct rme96 *rme96 = card->private_data; pci_restore_state(pci); - pci_enable_device(pci); + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "rme96: pci_enable_device failed, disabling device\n"); + snd_card_disconnect(card); + return -EIO; + } /* reset playback and record buffer pointers */ writel(0, rme96->iobase + RME96_IO_SET_PLAY_POS -- cgit v0.10.2 From eb63231830360f5acfea5dd2b545d7a14476bc3a Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Wed, 14 Aug 2013 12:27:33 +0200 Subject: ASoc: kirkwood: add DT support to the mvebu audio subsystem This patch adds DT support to the audio subsystem of the mvebu family (Kirkwood, Dove, Armada 370). Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt new file mode 100644 index 0000000..7e5fd37 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt @@ -0,0 +1,29 @@ +* mvebu (Kirkwood, Dove, Armada 370) audio controller + +Required properties: + +- compatible: "marvell,mvebu-audio" + +- reg: physical base address of the controller and length of memory mapped + region. + +- interrupts: list of two irq numbers. + The first irq is used for data flow and the second one is used for errors. + +- clocks: one or two phandles. + The first one is mandatory and defines the internal clock. + The second one is optional and defines an external clock. + +- clock-names: names associated to the clocks: + "internal" for the internal clock + "extclk" for the external clock + +Example: + +i2s1: audio-controller@b4000 { + compatible = "marvell,mvebu-audio"; + reg = <0xb4000 0x2210>; + interrupts = <21>, <22>; + clocks = <&gate_clk 13>; + clock-names = "internal"; +}; diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index e5f3f7a..7fce340 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -22,6 +22,8 @@ #include #include #include +#include + #include "kirkwood.h" #define DRV_NAME "mvebu-audio" @@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai; struct kirkwood_dma_data *priv; struct resource *mem; + struct device_node *np = pdev->dev.of_node; int err; priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); @@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) return -ENXIO; } - if (!data) { - dev_err(&pdev->dev, "no platform data ?!\n"); + if (np) { + priv->burst = 128; /* might be 32 or 128 */ + } else if (data) { + priv->burst = data->burst; + } else { + dev_err(&pdev->dev, "no DT nor platform data ?!\n"); return -EINVAL; } - priv->burst = data->burst; - - priv->clk = devm_clk_get(&pdev->dev, NULL); + priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL); if (IS_ERR(priv->clk)) { dev_err(&pdev->dev, "no clock\n"); return PTR_ERR(priv->clk); @@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev) priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24; /* Select the burst size */ - if (data->burst == 32) { + if (priv->burst == 32) { priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32; priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32; } else { @@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) return 0; } +#ifdef CONFIG_OF +static struct of_device_id mvebu_audio_of_match[] = { + { .compatible = "marvell,mvebu-audio" }, + { } +}; +MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); +#endif + static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .owner = THIS_MODULE, + .of_match_table = of_match_ptr(mvebu_audio_of_match), }, }; -- cgit v0.10.2 From a8cc20999799a94929a56393ff39b32245e33d64 Mon Sep 17 00:00:00 2001 From: Knut Petersen Date: Thu, 22 Aug 2013 14:36:16 +0200 Subject: alsa/rme96: Add missing inclusion of linux/vmalloc.h Signed-off-by: Knut Petersen Signed-off-by: Takashi Iwai diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 9d2a81f..bb9ebc5 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include -- cgit v0.10.2 From c445be35956b0cefe85db75d1e7994af5cecf16a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 23 Aug 2013 14:35:17 -0300 Subject: ASoC: simple-card: Provide owner and MODULE_ALIAS() Add .owner field and also MODULE_ALIAS(), so that auto module loading can work. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 6cf8355..8c49147 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev) static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .owner = THIS_MODULE, }, .probe = asoc_simple_card_probe, .remove = asoc_simple_card_remove, @@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = { module_platform_driver(asoc_simple_card); +MODULE_ALIAS("platform:asoc-simple-card"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("ASoC Simple Sound Card"); MODULE_AUTHOR("Kuninori Morimoto "); -- cgit v0.10.2 From 5af407cd365c8aab8a20e66aa6e4bc4a4983979e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 23 Aug 2013 18:14:45 -0300 Subject: ASoC: fsl_spdif: Remove unnecessary dev_set_drvdata() Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it from here. Signed-off-by: Fabio Estevam Reviewed-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 42a4382..a8ef46a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1184,7 +1184,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) &spdif_priv->cpu_dai_drv, 1); if (ret) { dev_err(&pdev->dev, "failed to register DAI: %d\n", ret); - goto error_dev; + return ret; } ret = imx_pcm_dma_init(pdev); @@ -1197,8 +1197,6 @@ static int fsl_spdif_probe(struct platform_device *pdev) error_component: snd_soc_unregister_component(&pdev->dev); -error_dev: - dev_set_drvdata(&pdev->dev, NULL); return ret; } @@ -1207,7 +1205,6 @@ static int fsl_spdif_remove(struct platform_device *pdev) { imx_pcm_dma_exit(pdev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); return 0; } -- cgit v0.10.2 From 6d22db43cf8b841dae37e7e3ee284c2b6c91a58b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Fri, 23 Aug 2013 18:14:46 -0300 Subject: ASoC: fsl_spdif: Reduce the noise on comments Remove the "====" pattern to let the comments cleaner and more uniform. Also, do not use multi-line style for a single line comment. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index a8ef46a..a9798aa 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -555,7 +555,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = { /* - * ============================================ * FSL SPDIF IEC958 controller(mixer) functions * * Channel status get/put control @@ -563,7 +562,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = { * Valid bit value get control * DPLL lock status get control * User bit sync mode selection control - * ============================================ */ static int fsl_spdif_info(struct snd_kcontrol *kcontrol, @@ -942,11 +940,7 @@ static const struct snd_soc_component_driver fsl_spdif_component = { .name = "fsl-spdif", }; -/* - * ================ - * FSL SPDIF REGMAP - * ================ - */ +/* FSL SPDIF REGMAP */ static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg) { -- cgit v0.10.2 From 53110a256a334c5e01db2d94c5306b4880a9180e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Sun, 25 Aug 2013 23:36:23 -0700 Subject: ASoC: fsi: reserve prefetch period on DMA transferring Current FSI is supporting DMAEngine transfer, but, it needs to use work queue. Therefore, DMA transfer settings might be late if there is heavy task. This patch reserves next period beforehand on DMA transfer function. Android sound will be breaking up without this patch. Tested-by: Tomohito Esaki Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 3039026..b33ca7c 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -235,6 +235,8 @@ struct fsi_stream { struct sh_dmae_slave slave; /* see fsi_handler_init() */ struct work_struct work; dma_addr_t dma; + int loop_cnt; + int additional_pos; }; struct fsi_clk { @@ -1289,6 +1291,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io) io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) | BUSOP_SET(16, PACKAGE_16BITBUS_STREAM); + io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */ + io->additional_pos = 0; io->dma = dma_map_single(dai->dev, runtime->dma_area, snd_pcm_lib_buffer_bytes(io->substream), dir); return 0; @@ -1305,11 +1309,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io) return 0; } -static dma_addr_t fsi_dma_get_area(struct fsi_stream *io) +static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional) { struct snd_pcm_runtime *runtime = io->substream->runtime; + int period = io->period_pos + additional; - return io->dma + samples_to_bytes(runtime, io->buff_sample_pos); + if (period >= runtime->periods) + period = 0; + + return io->dma + samples_to_bytes(runtime, period * io->period_samples); } static void fsi_dma_complete(void *data) @@ -1321,7 +1329,7 @@ static void fsi_dma_complete(void *data) enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ? DMA_TO_DEVICE : DMA_FROM_DEVICE; - dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io), + dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0), samples_to_bytes(runtime, io->period_samples), dir); io->buff_sample_pos += io->period_samples; @@ -1347,7 +1355,7 @@ static void fsi_dma_do_work(struct work_struct *work) struct snd_pcm_runtime *runtime; enum dma_data_direction dir; int is_play = fsi_stream_is_play(fsi, io); - int len; + int len, i; dma_addr_t buf; if (!fsi_stream_is_working(fsi, io)) @@ -1357,26 +1365,33 @@ static void fsi_dma_do_work(struct work_struct *work) runtime = io->substream->runtime; dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; len = samples_to_bytes(runtime, io->period_samples); - buf = fsi_dma_get_area(io); - dma_sync_single_for_device(dai->dev, buf, len, dir); + for (i = 0; i < io->loop_cnt; i++) { + buf = fsi_dma_get_area(io, io->additional_pos); - desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - if (!desc) { - dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); - return; - } + dma_sync_single_for_device(dai->dev, buf, len, dir); - desc->callback = fsi_dma_complete; - desc->callback_param = io; + desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + if (!desc) { + dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); + return; + } - if (dmaengine_submit(desc) < 0) { - dev_err(dai->dev, "tx_submit() fail\n"); - return; + desc->callback = fsi_dma_complete; + desc->callback_param = io; + + if (dmaengine_submit(desc) < 0) { + dev_err(dai->dev, "tx_submit() fail\n"); + return; + } + + dma_async_issue_pending(io->chan); + + io->additional_pos = 1; } - dma_async_issue_pending(io->chan); + io->loop_cnt = 1; /* * FIXME -- cgit v0.10.2 From f61df384282dfd1ca845e73ca8b8a187b87eb38a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 26 Aug 2013 09:25:13 -0300 Subject: ASoC: fsl_ssi: Remove unnecessary dev_set_drvdata() Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it from here. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 5cf626c..c6b7439 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1114,7 +1114,6 @@ error_dai: snd_soc_unregister_component(&pdev->dev); error_dev: - dev_set_drvdata(&pdev->dev, NULL); device_remove_file(&pdev->dev, dev_attr); error_clk: -- cgit v0.10.2 From f1aa06847506d5b88f5eb41fae6a24a7128097e7 Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Mon, 26 Aug 2013 21:35:21 -0400 Subject: ALSA: hda - add flags and routines to get devices selection info for DP1.2 MST This patch adds flags and routines to get device list & selection info on a pin. To support Display Port 1.2 multi-stream transport (MST) over single DP port, a pin can support multiple devices. Please refer to HD-A spec Document Change Notificaton HDA040-A. A display audio codec can set flag "dp_mst" in its patch, indicating its pins can support MST. But at runtime, a pin may not be multi-streaming capable and report the device list is empty, depending on Gfx driver configuration. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index fdbb09a..5b6c4e3 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -666,6 +666,64 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, } EXPORT_SYMBOL_HDA(snd_hda_get_conn_index); + +/* return DEVLIST_LEN parameter of the given widget */ +static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int parm; + + if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) || + get_wcaps_type(wcaps) != AC_WID_PIN) + return 0; + + parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN); + if (parm == -1 && codec->bus->rirb_error) + parm = 0; + return parm & AC_DEV_LIST_LEN_MASK; +} + +/** + * snd_hda_get_devices - copy device list without cache + * @codec: the HDA codec + * @nid: NID of the pin to parse + * @dev_list: device list array + * @max_devices: max. number of devices to store + * + * Copy the device list. This info is dynamic and so not cached. + * Currently called only from hda_proc.c, so not exported. + */ +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + u8 *dev_list, int max_devices) +{ + unsigned int parm; + int i, dev_len, devices; + + parm = get_num_devices(codec, nid); + if (!parm) /* not multi-stream capable */ + return 0; + + dev_len = parm + 1; + dev_len = dev_len < max_devices ? dev_len : max_devices; + + devices = 0; + while (devices < dev_len) { + parm = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_LIST, devices); + if (parm == -1 && codec->bus->rirb_error) + break; + + for (i = 0; i < 8; i++) { + dev_list[devices] = (u8)parm; + parm >>= 4; + devices++; + if (devices >= dev_len) + break; + } + } + return devices; +} + /** * snd_hda_queue_unsol_event - add an unsolicited event to queue * @bus: the BUS diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 701c2e0..b838c70 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -94,6 +94,8 @@ enum { #define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 #define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 #define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 +#define AC_VERB_GET_DEVICE_SEL 0xf35 +#define AC_VERB_GET_DEVICE_LIST 0xf36 /* * SET verbs @@ -133,6 +135,7 @@ enum { #define AC_VERB_SET_HDMI_DIP_XMIT 0x732 #define AC_VERB_SET_HDMI_CP_CTRL 0x733 #define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 +#define AC_VERB_SET_DEVICE_SEL 0x735 /* * Parameter IDs @@ -154,6 +157,7 @@ enum { #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 #define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_DEVLIST_LEN 0x15 #define AC_PAR_HDMI_LPCM_CAP 0x20 /* @@ -352,6 +356,10 @@ enum { #define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */ #define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */ +/* Display pin's device list length */ +#define AC_DEV_LIST_LEN_MASK 0x3f +#define AC_MAX_DEV_LIST_LEN 64 + /* * Control Parameters */ @@ -460,6 +468,11 @@ enum { #define AC_DEFCFG_PORT_CONN (0x3<<30) #define AC_DEFCFG_PORT_CONN_SHIFT 30 +/* Display pin's device list entry */ +#define AC_DE_PD (1<<0) +#define AC_DE_ELDV (1<<1) +#define AC_DE_IA (1<<2) + /* device device types (0x0-0xf) */ enum { AC_JACK_LINE_OUT, @@ -885,6 +898,7 @@ struct hda_codec { unsigned int pcm_format_first:1; /* PCM format must be set first */ unsigned int epss:1; /* supporting EPSS? */ unsigned int cached_write:1; /* write only to caches */ + unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */ #ifdef CONFIG_PM unsigned int power_on :1; /* current (global) power-state */ unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ @@ -972,6 +986,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums, const hda_nid_t *list); int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux, hda_nid_t nid, int recursive); +int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid, + u8 *dev_list, int max_devices); int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp); -- cgit v0.10.2 From 7a624ea56222fc6f6e3ccd135efedc195ba0b28d Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Mon, 26 Aug 2013 21:35:31 -0400 Subject: ALSA: hda - add device list & select info of display pins to codec proc file If a display codec supports multi-stream transport on the pins, the pin's device list length and device entries will be exposed to codec proc file. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 9760f00..a8cb22e 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -582,6 +582,36 @@ static void print_gpio(struct snd_info_buffer *buffer, print_nid_array(buffer, codec, nid, &codec->nids); } +static void print_device_list(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid) +{ + int i, curr = -1; + u8 dev_list[AC_MAX_DEV_LIST_LEN]; + int devlist_len; + + devlist_len = snd_hda_get_devices(codec, nid, dev_list, + AC_MAX_DEV_LIST_LEN); + snd_iprintf(buffer, " Devices: %d\n", devlist_len); + if (devlist_len <= 0) + return; + + curr = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DEVICE_SEL, 0); + + for (i = 0; i < devlist_len; i++) { + if (i == curr) + snd_iprintf(buffer, " *"); + else + snd_iprintf(buffer, " "); + + snd_iprintf(buffer, + "Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i, + !!(dev_list[i] & AC_DE_PD), + !!(dev_list[i] & AC_DE_ELDV), + !!(dev_list[i] & AC_DE_IA)); + } +} + static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { @@ -751,6 +781,9 @@ static void print_codec_info(struct snd_info_entry *entry, (wid_caps & AC_WCAP_DELAY) >> AC_WCAP_DELAY_SHIFT); + if (wid_type == AC_WID_PIN && codec->dp_mst) + print_device_list(buffer, codec, nid); + if (wid_caps & AC_WCAP_CONN_LIST) print_conn_list(buffer, codec, nid, wid_type, conn, conn_len); -- cgit v0.10.2 From 5dc989bdd968f369fec47d25343868ff9702953a Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Mon, 26 Aug 2013 21:35:41 -0400 Subject: ALSA: hda - Haswell codec exposes device list/select info on pins This patch is only to allow codec proc file to expose devices list/select info for Haswell codec pins. Since Haswell Gfx driver cannot support DP1.2 MST now, so all pins' device list is empty, meaning no pin is multi-streaming capaple. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 030ca86..87ca984 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1989,8 +1989,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; - if (codec->vendor_id == 0x80862807) + if (codec->vendor_id == 0x80862807) { codec->patch_ops.set_power_state = haswell_set_power_state; + codec->dp_mst = true; + } generic_hdmi_init_per_pins(codec); -- cgit v0.10.2 From 2e59e5ab1c24489c5581b83e56b0435432c54dfe Mon Sep 17 00:00:00 2001 From: Mengdong Lin Date: Mon, 26 Aug 2013 21:35:49 -0400 Subject: ALSA: hda - add device entry and inactive flag to unsolicited response This patch adds two fields to unsolicited response, according to spec HDA040-A: - Device Entry (bit 20:15) - Inactive (bit 2) and show the info in debug message. Signed-off-by: Mengdong Lin Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index b838c70..7aa9870 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -255,6 +255,11 @@ enum { #define AC_UNSOL_RES_TAG_SHIFT 26 #define AC_UNSOL_RES_SUBTAG (0x1f<<21) #define AC_UNSOL_RES_SUBTAG_SHIFT 21 +#define AC_UNSOL_RES_DE (0x3f<<15) /* Device Entry + * (for DP1.2 MST) + */ +#define AC_UNSOL_RES_DE_SHIFT 15 +#define AC_UNSOL_RES_IA (1<<2) /* Inactive (for DP1.2 MST) */ #define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ #define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ #define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 87ca984..895a0d3 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -959,6 +959,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) int pin_nid; int pin_idx; struct hda_jack_tbl *jack; + int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) @@ -967,8 +968,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, - "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, + "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", + codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); -- cgit v0.10.2 From 1c9a341bbdc14051a4d8c74ea67269786c7d3736 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 27 Aug 2013 14:49:59 +0200 Subject: ALSA: hda - Simplify CONFIG_SND_HDA_I915 condition CONFIG_SND_HDA_I915 doesn't have to be user-selectable as this is almost mandatory when i915 driver is available. Let's enable it always when CONFIG_DRM_I915 is set, so that user won't be bothered by useless questions. Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 59c5e9c..8de66cc 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -152,14 +152,9 @@ config SND_HDA_CODEC_HDMI This module is automatically loaded at probing. config SND_HDA_I915 - bool "Build Display HD-audio controller/codec power well support for i915 cards" + bool + default y depends on DRM_I915 - help - Say Y here to include full HDMI and DisplayPort HD-audio controller/codec - power-well support for Intel Haswell graphics cards based on the i915 driver. - - Note that this option must be enabled for Intel Haswell C+ stepping machines, otherwise - the GPU audio controller/codecs will not be initialized or damaged when exit from S3 mode. config SND_HDA_CODEC_CIRRUS bool "Build Cirrus Logic codec support" -- cgit v0.10.2 From a85f9da707366e856c0aad9e329db0cc59475290 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Aug 2013 15:50:55 +0200 Subject: ASoC: dmic: Convert table based DAPM setup Let the core take care of instantiating the DAPM widgets and routes, this makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c index 66967ba..b2090b2 100644 --- a/sound/soc/codecs/dmic.c +++ b/sound/soc/codecs/dmic.c @@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = { {"DMIC AIF", NULL, "DMic"}, }; -static int dmic_probe(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets, - ARRAY_SIZE(dmic_dapm_widgets)); - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - snd_soc_dapm_new_widgets(dapm); - - return 0; -} - static struct snd_soc_codec_driver soc_dmic = { - .probe = dmic_probe, + .dapm_widgets = dmic_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int dmic_dev_probe(struct platform_device *pdev) -- cgit v0.10.2 From 34742cb02bd368c1af3349c041d3e4446f7ac6ef Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Aug 2013 15:50:54 +0200 Subject: ASoC: dapm: Fix marking widgets dirty when a route is added The current calls to dapm_mark_dirty() in snd_soc_dapm_add_path() are on a path that is only reached if the sink widget is either a mixer or a mux. Move the calls further up so they are called for all widget types. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d84bd0f..7e9afbc4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2374,6 +2374,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, wsource->ext = 1; } + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + /* connect static paths */ if (control == NULL) { list_add(&path->list, &dapm->card->paths); @@ -2436,9 +2439,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, return 0; } - dapm_mark_dirty(wsource, "Route added"); - dapm_mark_dirty(wsink, "Route added"); - return 0; err: kfree(path); -- cgit v0.10.2 From aac97b5fd9537b62a68830d189509297cdac5ad9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Aug 2013 15:50:56 +0200 Subject: ASoC: tlv320aic32x4: Convert table based control and DAPM setup Let the core take care of instantiating the controls and DAPM widgets and routes, this makes the code a bit shorter. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 17df4e3..2ed57d4 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -338,18 +338,6 @@ static inline int aic32x4_get_divs(int mclk, int rate) return -EINVAL; } -static int aic32x4_add_widgets(struct snd_soc_codec *codec) -{ - snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets, - ARRAY_SIZE(aic32x4_dapm_widgets)); - - snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes, - ARRAY_SIZE(aic32x4_dapm_routes)); - - snd_soc_dapm_new_widgets(&codec->dapm); - return 0; -} - static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -683,9 +671,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec) } aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - snd_soc_add_codec_controls(codec, aic32x4_snd_controls, - ARRAY_SIZE(aic32x4_snd_controls)); - aic32x4_add_widgets(codec); /* * Workaround: for an unknown reason, the ADC needs to be powered up @@ -714,6 +699,13 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { .suspend = aic32x4_suspend, .resume = aic32x4_resume, .set_bias_level = aic32x4_set_bias_level, + + .controls = aic32x4_snd_controls, + .num_controls = ARRAY_SIZE(aic32x4_snd_controls), + .dapm_widgets = aic32x4_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), + .dapm_routes = aic32x4_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; static int aic32x4_i2c_probe(struct i2c_client *i2c, -- cgit v0.10.2 From 318ee162c882526685be4f44d7b519cdcc45cbfe Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Aug 2013 15:50:57 +0200 Subject: ASoC: wm8904: Remove unnecessary call to snd_soc_dapm_new_widgets() The core will call snd_soc_dapm_new_widgets() once all components of the card have been initialized, so there is no need to do this manually in the driver. Calling it earlier also might result in a partially instantiated system being powered up which cause undesired side effects. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 91dfbfe..4dfa8dc 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1202,7 +1202,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec) break; } - snd_soc_dapm_new_widgets(dapm); return 0; } -- cgit v0.10.2 From 148663074c1778d88c9e9c5f5cc66493ed30fa25 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Aug 2013 15:50:58 +0200 Subject: ASoC: jack: Remove unnecessary call to snd_soc_dapm_new_widgets() snd_soc_jack_add_pins() does not create any new DAPM widgets, so there is no need to call snd_soc_dapm_new_widgets(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 7aa26b5..71358e3 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -183,8 +183,6 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count, list_add(&(pins[i].list), &jack->pins); } - snd_soc_dapm_new_widgets(&jack->codec->card->dapm); - /* Update to reflect the last reported status; canned jack * implementations are likely to set their state before the * card has an opportunity to associate pins. -- cgit v0.10.2 From 4b52fa211a7c65eab78acf3f434361d40de87688 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Aug 2013 15:50:59 +0200 Subject: ASoC: Call snd_soc_dapm_new_widgets() only once during card initialization Each time snd_soc_dapm_new_widgets() is called it will instantiate all the widgets and routes that have been added so far and then power them. Doing this multiple times before the card is fully initialized and all widgets have been added can cause unnecessary and even invalid power state transitions which can result in extra register writes and and also might cause clicks and pops. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f46472d..85e2a8b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1230,9 +1230,6 @@ static int soc_post_component_init(struct snd_soc_card *card, } rtd->card = card; - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(&codec->dapm); - /* machine controls, routes and widgets are not prefixed */ temp = codec->name_prefix; codec->name_prefix = NULL; @@ -1728,8 +1725,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); - snd_soc_dapm_new_widgets(&card->dapm); - for (i = 0; i < card->num_links; i++) { dai_link = &card->dai_link[i]; dai_fmt = dai_link->dai_fmt; -- cgit v0.10.2 From 8c193b8dce4f2a2474dc2bc39ec972454df9d439 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Aug 2013 15:51:00 +0200 Subject: ASoC: Move call to snd_soc_dapm_new_widgets() after snd_soc_dapm_auto_nc_codec_pins() Call snd_soc_dapm_new_widgets() before the auto non-connected pins have been marked as not connected will power the system under the assumption that those pins are connected. Once the pins have been marked as disconnected the system there will be an additional power run. This can cause unnecessary power transitions. Calling snd_soc_dapm_new_widgets() only after the pins have been marked as non-connected avoids this. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 85e2a8b..9375012 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1803,12 +1803,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) } } - snd_soc_dapm_new_widgets(&card->dapm); - if (card->fully_routed) list_for_each_entry(codec, &card->codec_dev_list, card_list) snd_soc_dapm_auto_nc_codec_pins(codec); + snd_soc_dapm_new_widgets(&card->dapm); + ret = snd_card_register(card->snd_card); if (ret < 0) { dev_err(card->dev, "ASoC: failed to register soundcard %d\n", -- cgit v0.10.2 From 824ef826f3c4d83d1925a5e351313bfd3e5ca6cb Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 27 Aug 2013 15:51:01 +0200 Subject: ASoC: Pass card instead of dapm context to snd_soc_dapm_new_widgets() snd_soc_dapm_new_widgets() works on the ASoC card as a whole not on a specific DAPM context. The DAPM context that is passed as the parameter is only used to look up the pointer to the card. This patch updates the signature of snd_soc_dapm_new_widgets() to take the card directly. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index c728d28..27a72d5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -413,7 +413,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, struct snd_soc_dapm_widget *sink); /* dapm path setup */ -int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm); +int snd_soc_dapm_new_widgets(struct snd_soc_card *card); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm); int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d476f75..ed3c253 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1807,7 +1807,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) list_for_each_entry(codec, &card->codec_dev_list, card_list) snd_soc_dapm_auto_nc_codec_pins(codec); - snd_soc_dapm_new_widgets(&card->dapm); + snd_soc_dapm_new_widgets(card); ret = snd_card_register(card->snd_card); if (ret < 0) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e9afbc4..548b1c9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2712,9 +2712,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); * * Returns 0 for success. */ -int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) +int snd_soc_dapm_new_widgets(struct snd_soc_card *card) { - struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; unsigned int val; -- cgit v0.10.2 From 446a3bd4329bcaf95d71c6717c2c424a0f97ff18 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Tue, 27 Aug 2013 20:27:11 +0200 Subject: ASoc: kirkwood: Use the Kirkwood audio driver in Dove boards This patch permits the generation of the Kirkwood audio driver which may be used in the Dove boards. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 9e1970c..78ed4a4 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,6 @@ config SND_KIRKWOOD_SOC - tristate "SoC Audio for the Marvell Kirkwood chip" - depends on ARCH_KIRKWOOD || COMPILE_TEST + tristate "SoC Audio for the Marvell Kirkwood and Dove chips" + depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- cgit v0.10.2 From 2a956ec04b3703809b6cf500dbee450e44f3a70c Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 28 Aug 2013 12:04:46 +0800 Subject: ASoC: fsl: Add S/PDIF machine driver This patch implements a device-tree-only machine driver for Freescale i.MX series Soc. It works with spdif_transmitter/spdif_receiver and fsl_spdif.c drivers. Signed-off-by: Nicolin Chen Acked-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt new file mode 100644 index 0000000..7d13479 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt @@ -0,0 +1,34 @@ +Freescale i.MX audio complex with S/PDIF transceiver + +Required properties: + + - compatible : "fsl,imx-audio-spdif" + + - model : The user-visible name of this sound complex + + - spdif-controller : The phandle of the i.MX S/PDIF controller + + +Optional properties: + + - spdif-out : This is a boolean property. If present, the transmitting + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF out connector/jack on the board or it's connecting to some + other IP block, such as an HDMI encoder/display-controller. + + - spdif-in : This is a boolean property. If present, the receiving + function of S/PDIF will be enabled, indicating there's a physical + S/PDIF in connector/jack on the board. + +* Note: At least one of these two properties should be set in the DT binding. + + +Example: + +sound-spdif { + compatible = "fsl,imx-audio-spdif"; + model = "imx-spdif"; + spdif-controller = <&spdif>; + spdif-out; + spdif-in; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index cd088cc..a708380 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000 Say Y if you want to add support for SoC audio on an i.MX board with a sgtl5000 codec. +config SND_SOC_IMX_SPDIF + tristate "SoC Audio support for i.MX boards with S/PDIF" + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SPDIF + select SND_SOC_FSL_UTILS + select SND_SOC_SPDIF + help + SoC Audio support for i.MX boards with S/PDIF + Say Y if you want to add support for SoC audio on an i.MX board with + a S/DPDIF. + config SND_SOC_IMX_MC13783 tristate "SoC Audio support for I.MX boards with mc13783" depends on MFD_MC13783 && ARM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 4b5970e..e2aaff7 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o +snd-soc-imx-spdif-objs :=imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o +obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c new file mode 100644 index 0000000..816013b --- /dev/null +++ b/sound/soc/fsl/imx-spdif.c @@ -0,0 +1,148 @@ +/* + * Copyright (C) 2013 Freescale Semiconductor, Inc. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include +#include +#include + +struct imx_spdif_data { + struct snd_soc_dai_link dai[2]; + struct snd_soc_card card; + struct platform_device *txdev; + struct platform_device *rxdev; +}; + +static int imx_spdif_audio_probe(struct platform_device *pdev) +{ + struct device_node *spdif_np, *np = pdev->dev.of_node; + struct imx_spdif_data *data; + int ret = 0, num_links = 0; + + spdif_np = of_parse_phandle(np, "spdif-controller", 0); + if (!spdif_np) { + dev_err(&pdev->dev, "failed to find spdif-controller\n"); + ret = -EINVAL; + goto end; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + dev_err(&pdev->dev, "failed to allocate memory\n"); + ret = -ENOMEM; + goto end; + } + + if (of_property_read_bool(np, "spdif-out")) { + data->dai[num_links].name = "S/PDIF TX"; + data->dai[num_links].stream_name = "S/PDIF PCM Playback"; + data->dai[num_links].codec_dai_name = "dit-hifi"; + data->dai[num_links].codec_name = "spdif-dit"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0); + if (IS_ERR(data->txdev)) { + ret = PTR_ERR(data->txdev); + dev_err(&pdev->dev, "register dit failed: %d\n", ret); + goto end; + } + } + + if (of_property_read_bool(np, "spdif-in")) { + data->dai[num_links].name = "S/PDIF RX"; + data->dai[num_links].stream_name = "S/PDIF PCM Capture"; + data->dai[num_links].codec_dai_name = "dir-hifi"; + data->dai[num_links].codec_name = "spdif-dir"; + data->dai[num_links].cpu_of_node = spdif_np; + data->dai[num_links].platform_of_node = spdif_np; + num_links++; + + data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0); + if (IS_ERR(data->rxdev)) { + ret = PTR_ERR(data->rxdev); + dev_err(&pdev->dev, "register dir failed: %d\n", ret); + goto error_dit; + } + } + + if (!num_links) { + dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n"); + goto error_dir; + } + + data->card.dev = &pdev->dev; + data->card.num_links = num_links; + data->card.dai_link = data->dai; + + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto error_dir; + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret); + goto error_dir; + } + + platform_set_drvdata(pdev, data); + + goto end; + +error_dir: + if (data->rxdev) + platform_device_unregister(data->rxdev); +error_dit: + if (data->txdev) + platform_device_unregister(data->txdev); +end: + if (spdif_np) + of_node_put(spdif_np); + + return ret; +} + +static int imx_spdif_audio_remove(struct platform_device *pdev) +{ + struct imx_spdif_data *data = platform_get_drvdata(pdev); + + if (data->rxdev) + platform_device_unregister(data->rxdev); + if (data->txdev) + platform_device_unregister(data->txdev); + + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_spdif_dt_ids[] = { + { .compatible = "fsl,imx-audio-spdif", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids); + +static struct platform_driver imx_spdif_driver = { + .driver = { + .name = "imx-spdif", + .owner = THIS_MODULE, + .of_match_table = imx_spdif_dt_ids, + }, + .probe = imx_spdif_audio_probe, + .remove = imx_spdif_audio_remove, +}; + +module_platform_driver(imx_spdif_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-spdif"); -- cgit v0.10.2 From bfd7d1aa3b603cf43e6545f873de714b991d6a8a Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Thu, 29 Aug 2013 08:00:05 +0800 Subject: ASoC: fsl_spdif: remove redundant dev_err call in fsl_spdif_probe() There is a error message within devm_ioremap_resource already, so remove the dev_err call to avoid redundant error message. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index a9798aa..e93dc0d 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1113,10 +1113,8 @@ static int fsl_spdif_probe(struct platform_device *pdev) } regs = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(regs)) { - dev_err(&pdev->dev, "could not map device resources\n"); + if (IS_ERR(regs)) return PTR_ERR(regs); - } spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "core", regs, &fsl_spdif_regmap_config); -- cgit v0.10.2 From e925a6b1b6e7ddb43a71b31c0afa12ca9a6ec118 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 26 Aug 2013 09:25:15 -0300 Subject: ASoC: designware_i2s: Remove unnecessary dev_set_drvdata() Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it from here. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 70eb37a..25c31f1 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -421,13 +421,11 @@ static int dw_i2s_probe(struct platform_device *pdev) dw_i2s_dai, 1); if (ret != 0) { dev_err(&pdev->dev, "not able to register dai\n"); - goto err_set_drvdata; + goto err_clk_disable; } return 0; -err_set_drvdata: - dev_set_drvdata(&pdev->dev, NULL); err_clk_disable: clk_disable(dev->clk); err_clk_put: @@ -440,7 +438,6 @@ static int dw_i2s_remove(struct platform_device *pdev) struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); clk_put(dev->clk); -- cgit v0.10.2 From ba1fb69508615011eba225de1ed2615fa205be9a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Mon, 26 Aug 2013 09:25:14 -0300 Subject: ASoC: ep93xx-i2s: Remove unnecessary dev_set_drvdata() Driver core clears the driver data to NULL after device_release or on probe failure, so just remove it from here. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index f23f331..a57643d 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -408,7 +408,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev) return 0; fail_put_lrclk: - dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); fail_put_sclk: clk_put(info->sclk); @@ -423,7 +422,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev) struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev); snd_soc_unregister_component(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); clk_put(info->lrclk); clk_put(info->sclk); clk_put(info->mclk); -- cgit v0.10.2 From 9b9ae16a97e08bdc4fd5e726a4d17119dbae5d8a Mon Sep 17 00:00:00 2001 From: Tomasz Figa Date: Sun, 11 Aug 2013 19:59:21 +0200 Subject: ASoC: Samsung: Do not queue cyclic buffers multiple times The legacy S3C-DMA API required every period of a cyclic buffer to be queued separately. After conversion of Samsung ASoC to Samsung DMA wrappers somebody made an assumption that the same is needed for DMA engine API, which is not true. In effect, Samsung ASoC DMA code was queuing the whole cyclic buffer multiple times with a shift of one period per iteration, leading to: a) severe memory waste - up to 13x times more DMA transfer descriptors are allocated than needed, b) possible memory corruption, because further cyclic buffers were out of the original buffers, due to the offset. This patch fixes this problem by making the legacy S3C-DMA API use the same semantics as DMA engine (the whole cyclic buffer is enqueued at once) and modifying users of Samsung DMA wrappers in cyclic mode to behave appropriately. Signed-off-by: Tomasz Figa Acked-by: Linus Walleij Signed-off-by: Mark Brown diff --git a/arch/arm/plat-samsung/s3c-dma-ops.c b/arch/arm/plat-samsung/s3c-dma-ops.c index 0cc40ae..98b10ba 100644 --- a/arch/arm/plat-samsung/s3c-dma-ops.c +++ b/arch/arm/plat-samsung/s3c-dma-ops.c @@ -82,7 +82,8 @@ static int s3c_dma_config(unsigned ch, struct samsung_dma_config *param) static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param) { struct cb_data *data; - int len = (param->cap == DMA_CYCLIC) ? param->period : param->len; + dma_addr_t pos = param->buf; + dma_addr_t end = param->buf + param->len; list_for_each_entry(data, &dma_list, node) if (data->ch == ch) @@ -94,7 +95,15 @@ static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param) data->fp_param = param->fp_param; } - s3c2410_dma_enqueue(ch, (void *)data, param->buf, len); + if (param->cap != DMA_CYCLIC) { + s3c2410_dma_enqueue(ch, (void *)data, param->buf, param->len); + return 0; + } + + while (pos < end) { + s3c2410_dma_enqueue(ch, (void *)data, pos, param->period); + pos += param->period; + } return 0; } diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index a0c67f6..9338d11 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -90,6 +90,13 @@ static void dma_enqueue(struct snd_pcm_substream *substream) dma_info.period = prtd->dma_period; dma_info.len = prtd->dma_period*limit; + if (dma_info.cap == DMA_CYCLIC) { + dma_info.buf = pos; + prtd->params->ops->prepare(prtd->params->ch, &dma_info); + prtd->dma_loaded += limit; + return; + } + while (prtd->dma_loaded < limit) { pr_debug("dma_loaded: %d\n", prtd->dma_loaded); -- cgit v0.10.2 From 2f82cdbafd53a01e3a3995a618b650653eed9c1a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 29 Aug 2013 17:31:41 -0300 Subject: ASoC: fsl: Drop SND_SOC_FSL_UTILS from SND_SOC_IMX_SPDIF SND_SOC_FSL_UTILS is only used by PowerPC machines, so let's drop it in the i.mx case. Signed-off-by: Fabio Estevam Acked-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index a708380..704e246 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -197,7 +197,6 @@ config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select SND_SOC_FSL_UTILS select SND_SOC_SPDIF help SoC Audio support for i.MX boards with S/PDIF -- cgit v0.10.2 From 2daabd7848b89afddd93be616f1be5639ea78822 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 30 Aug 2013 17:39:33 +0200 Subject: ASoC: dapm: Fix auto-disable for inverted controls We need to make sure that the control's cached value is initialized to the same value as the control's widget->on_val. Otherwise updates might be lost. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7e9afbc4..13fcb61 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -229,6 +229,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, template.id = snd_soc_dapm_kcontrol; template.name = kcontrol->id.name; + data->value = template.on_val; + data->widget = snd_soc_dapm_new_control(widget->dapm, &template); if (!data->widget) { -- cgit v0.10.2 From 246693ba7b0b824a970f9431486ad88c18e0ce2d Mon Sep 17 00:00:00 2001 From: Bard Liao Date: Fri, 23 Aug 2013 10:29:26 +0800 Subject: ASoC: rt5640: change widget sequence for depop Signed-off-by: Bard Liao Tested-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 4db7314..c26a8f8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = { static struct reg_default init_list[] = { {RT5640_PR_BASE + 0x3d, 0x3600}, - {RT5640_PR_BASE + 0x1c, 0x0D21}, - {RT5640_PR_BASE + 0x1b, 0x0000}, {RT5640_PR_BASE + 0x12, 0x0aa8}, {RT5640_PR_BASE + 0x14, 0x0aaa}, {RT5640_PR_BASE + 0x20, 0x6110}, @@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ - SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), /* Headphone Output Volume */ - SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL, - RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, @@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = { RT5640_M_BST1_MM_SFT, 1, 1), }; +static const struct snd_kcontrol_new spk_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL, + RT5640_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_enable_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL, + RT5640_R_MUTE_SFT, 1, 1); + /* Stereo ADC source */ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" @@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL( static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int spk_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - - switch (event) { - case SND_SOC_DAPM_POST_PMU: - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0001); - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0xf000); - break; - - case SND_SOC_DAPM_PRE_PMD: - regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c, - 0xf000, 0x0000); - regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1, - 0x0001, 0x0000); - break; - - default: - return 0; - } - return 0; -} - static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -943,6 +926,117 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, return 0; } +void hp_amp_power_on(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + /* depop parameters */ + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK, RT5640_DEPOP_MAN); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK, + RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU); + regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1, + 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2, 0); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_HA, + RT5640_PWR_HA); + usleep_range(10000, 15000); + regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1, + RT5640_PWR_FV1 | RT5640_PWR_FV2 , + RT5640_PWR_FV1 | RT5640_PWR_FV2); +} + +static void rt5640_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2, + RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK, + RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN); + regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP, + RT5640_PM_HP_MASK, RT5640_PM_HP_HV); + + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3, + RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK, + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) | + (RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) | + (RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT)); + + regmap_write(rt5640->regmap, RT5640_PR_BASE + + RT5640_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1, + RT5640_HP_CP_MASK | RT5640_HP_SG_MASK, + RT5640_HP_CP_PD | RT5640_HP_SG_EN); + regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + + RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400); +} + +static int rt5640_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5640_pmu_depop(codec); + rt5640->hp_mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5640->hp_mute = 1; + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power_on(codec); + break; + default: + return 0; + } + + return 0; +} + +static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5640->hp_mute) + usleep_range(80000, 85000); + + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2, RT5640_PWR_PLL_BIT, 0, NULL, 0), @@ -1132,15 +1226,28 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1, - SND_SOC_NOPM, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, + 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, + rt5640_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_L_BIT, 0, NULL, 0), - SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1, + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1, RT5640_PWR_HP_R_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1, - SND_SOC_NOPM, 0, spk_event, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + RT5640_PWR_CLS_D_BIT, 0, NULL, 0), + + /* Output Switch */ + SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0, + &spk_l_enable_control), + SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0, + &spk_r_enable_control), + SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0, + &hp_l_enable_control), + SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, + &hp_r_enable_control), + SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event), /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1381,9 +1488,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPO MIX L", NULL, "HP L Amp"}, {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, + {"HPO MIX R", NULL, "HP R Amp"}, {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, @@ -1396,13 +1505,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP L Amp", NULL, "HPO MIX L"}, - {"HP R Amp", NULL, "HPO MIX R"}, + {"HP Amp", NULL, "HPO MIX L"}, + {"HP Amp", NULL, "HPO MIX R"}, - {"SPOLP", NULL, "SPOL MIX"}, - {"SPOLN", NULL, "SPOL MIX"}, - {"SPORP", NULL, "SPOR MIX"}, - {"SPORN", NULL, "SPOR MIX"}, + {"Speaker L Playback", "Switch", "SPOL MIX"}, + {"Speaker R Playback", "Switch", "SPOR MIX"}, + {"SPOLP", NULL, "Speaker L Playback"}, + {"SPOLN", NULL, "Speaker L Playback"}, + {"SPORP", NULL, "Speaker R Playback"}, + {"SPORN", NULL, "Speaker R Playback"}, {"SPOLP", NULL, "Improve SPK Amp Drv"}, {"SPOLN", NULL, "Improve SPK Amp Drv"}, @@ -1412,8 +1523,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOL", NULL, "Improve HP Amp Drv"}, {"HPOR", NULL, "Improve HP Amp Drv"}, - {"HPOL", NULL, "HP L Amp"}, - {"HPOR", NULL, "HP R Amp"}, + {"HP L Playback", "Switch", "HP Amp"}, + {"HP R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HP L Playback"}, + {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, {"MONOP", NULL, "Mono MIX"}, @@ -1792,17 +1905,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, RT5640_PWR_BG | RT5640_PWR_VREF2, RT5640_PWR_VREF1 | RT5640_PWR_MB | RT5640_PWR_BG | RT5640_PWR_VREF2); - mdelay(10); + usleep_range(10000, 15000); snd_soc_update_bits(codec, RT5640_PWR_ANLG1, RT5640_PWR_FV1 | RT5640_PWR_FV2, RT5640_PWR_FV1 | RT5640_PWR_FV2); regcache_sync(rt5640->regmap); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, - 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, - 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); } @@ -1846,8 +1955,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); - snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019); - snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000); snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); @@ -2069,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + rt5640->hp_mute = 1; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, rt5640_dai, ARRAY_SIZE(rt5640_dai)); if (ret < 0) diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index c48286d..5e8df25a 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -145,6 +145,8 @@ /* Index of Codec Private Register definition */ +#define RT5640_CHPUMP_INT_REG1 0x24 +#define RT5640_MAMP_INT_REG2 0x37 #define RT5640_3D_SPK 0x63 #define RT5640_WND_1 0x6c #define RT5640_WND_2 0x6d @@ -153,6 +155,7 @@ #define RT5640_WND_5 0x70 #define RT5640_WND_8 0x73 #define RT5640_DIP_SPK_INF 0x75 +#define RT5640_HP_DCC_INT1 0x77 #define RT5640_EQ_BW_LOP 0xa0 #define RT5640_EQ_GN_LOP 0xa1 #define RT5640_EQ_FC_BP1 0xa2 @@ -1201,6 +1204,14 @@ #define RT5640_CP_FQ2_SFT 4 #define RT5640_CP_FQ3_MASK (0x7) #define RT5640_CP_FQ3_SFT 0 +#define RT5640_CP_FQ_1_5_KHZ 0 +#define RT5640_CP_FQ_3_KHZ 1 +#define RT5640_CP_FQ_6_KHZ 2 +#define RT5640_CP_FQ_12_KHZ 3 +#define RT5640_CP_FQ_24_KHZ 4 +#define RT5640_CP_FQ_48_KHZ 5 +#define RT5640_CP_FQ_96_KHZ 6 +#define RT5640_CP_FQ_192_KHZ 7 /* HPOUT charge pump (0x91) */ #define RT5640_OSW_L_MASK (0x1 << 11) @@ -2087,6 +2098,7 @@ struct rt5640_priv { int pll_out; int dmic_en; + bool hp_mute; }; #endif -- cgit v0.10.2 From 8a309d71ed9d17ff251b4b891fcef1c72bf625d1 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 30 Aug 2013 17:38:08 +0800 Subject: ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c Add wrapping '\n' for dev_dbg() in fsl_spdif.c Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e93dc0d..98741e9 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1071,9 +1071,9 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, break; } - dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate", + dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n", spdif_priv->txclk_src[index], rate[index]); - dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate", + dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n", spdif_priv->txclk_div[index], rate[index]); return 0; -- cgit v0.10.2 From 8626bdf05e93ae9d199cb2ad77b58832b98141f5 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 30 Aug 2013 17:39:00 +0800 Subject: ASoC: fsl: Add one blank space after ':=' in Makefile There is a blank space missing between ':=' and 'imx-spdif.o', thus add it. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index e2aaff7..8db705b 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -45,7 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o snd-soc-imx-wm8962-objs := imx-wm8962.o -snd-soc-imx-spdif-objs :=imx-spdif.o +snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o -- cgit v0.10.2 From 43d92e7d9aa13b91687f671ea7015204bc88fb84 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 11:02:17 +0200 Subject: ASoC: Remove unused control_type field from snd_soc_codec struct The control_type field was used by the core to track which raw IO methods to use, but when switching to regmap this was no longer necessary and so the last user of the field was removed in commit be3ea3b9 ("ASoC: Use new register map API for ASoC generic physical I/O"). The field is now completely unused and can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index d57a04e..254bcda 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -701,7 +701,6 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ - enum snd_soc_control_type control_type; hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); -- cgit v0.10.2 From ad758a67048f58205f2777b0f0a1a02de824d280 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 11:02:18 +0200 Subject: ASoC: Remove unused debugfs_dapm field from snd_soc_{platform,codec} struct The DAPM context struct has its own field where it stores the pointer to the DAPM debugfs entry. The debugfs_dapm field in the snd_soc_platform and snd_soc_codec structs are completely unused and can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index 254bcda..c65cc7d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -719,7 +719,6 @@ struct snd_soc_codec { #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_codec_root; struct dentry *debugfs_reg; - struct dentry *debugfs_dapm; #endif }; @@ -844,7 +843,6 @@ struct snd_soc_platform { #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_platform_root; - struct dentry *debugfs_dapm; #endif }; -- cgit v0.10.2 From d7b1538c7c0e395a308d6f4098d0985fe19e4584 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 11:02:19 +0200 Subject: ASoC: Remove unused sysfs_registered field from snd_soc_codec struct The sysfs_registered field was added to the snd_soc_codec struct in commit f0fba2ad ("ASoC: multi-component - ASoC Multi-Component Support"), but has never been used. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index c65cc7d..26e0df0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -693,7 +693,6 @@ struct snd_soc_codec { unsigned int probed:1; /* Codec has been probed */ unsigned int ac97_registered:1; /* Codec has been AC97 registered */ unsigned int ac97_created:1; /* Codec has been created by SoC */ - unsigned int sysfs_registered:1; /* codec has been sysfs registered */ unsigned int cache_init:1; /* codec cache has been initialized */ unsigned int using_regmap:1; /* using regmap access */ u32 cache_only; /* Suppress writes to hardware */ -- cgit v0.10.2 From 9d863b88ec371491e926e0828dbe3d36ead0f6f9 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 31 Aug 2013 18:15:23 +0200 Subject: ASoC: ssm2602: Fix cache sync The ssm2602 uses regmap for caching not soc-cache, so we need to use regcache_sync() instead of snd_soc_cache_sync(). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index f8d30e5..492644e 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec) static int ssm2602_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); + regcache_sync(ssm2602->regmap); ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- cgit v0.10.2 From 6b4c80f947df9d92b97eb62afc29dda6d7220c7d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 31 Aug 2013 16:40:51 +0100 Subject: ASoC: fsl_spdif: Staticse non-exported symbols Signed-off-by: Mark Brown Acked-by: Nicolin Chen diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 98741e9..3920c3e 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, return 0; } -int fsl_spdif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int fsl_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); @@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, return 0; } -struct snd_soc_dai_ops fsl_spdif_dai_ops = { +static struct snd_soc_dai_ops fsl_spdif_dai_ops = { .startup = fsl_spdif_startup, .hw_params = fsl_spdif_hw_params, .trigger = fsl_spdif_trigger, @@ -919,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) return 0; } -struct snd_soc_dai_driver fsl_spdif_dai = { +static struct snd_soc_dai_driver fsl_spdif_dai = { .probe = &fsl_spdif_dai_probe, .playback = { .channels_min = 2, -- cgit v0.10.2 From d6bead020d8f8bcaca5cdcb035250c44b21c93e7 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 29 Aug 2013 10:32:13 -0300 Subject: ASoC: soc-pcm: Allow to specify unidirectional dai_link Add 'playback_only' and 'capture_only' fields that can be used for specifying that a dai_link has a unidirectional capability. The motivation for this is for the cases of systems, such as Freescale MX28, that has two unidirectional DAIs. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index c0ac3bc..65414e8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -936,6 +936,10 @@ struct snd_soc_dai_link { /* machine stream operations */ const struct snd_soc_ops *ops; const struct snd_soc_compr_ops *compr_ops; + + /* For unidirectional dai links */ + bool playback_only; + bool capture_only; }; struct snd_soc_codec_conf { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b6c6403..9abaa52 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2027,6 +2027,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = 1; } + if (rtd->dai_link->playback_only) { + playback = 1; + capture = 0; + } + + if (rtd->dai_link->capture_only) { + playback = 0; + capture = 1; + } + /* create the PCM */ if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", -- cgit v0.10.2 From a90e6053baa61feed8b19a9f4cbec6b56479d1ba Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 29 Aug 2013 10:32:14 -0300 Subject: ASoC: mxs-sgtl5000: Configure the dai_links as unidirectional On a mx28 board, running "aplay -l" and "arecord -l" results in the following: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 ,which is not correct because we got a capture device listed in aplay and a playback device listed in arecord. On mx28 there are two serial audio interface ports (SAIF0 and SAIF1) and each one of them are unidirectional. Allow to specify a dai link as 'playback_only' or 'capture_only', which suits well for this case. After this change we can correctly report the capabilities as follows: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: HiFi Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 1: HiFi Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 Also tested playback and capture on the mx28evk board. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 1b134d7..ed8a519 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -91,11 +91,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = { .stream_name = "HiFi Playback", .codec_dai_name = "sgtl5000", .ops = &mxs_sgtl5000_hifi_ops, + .playback_only = true, }, { .name = "HiFi Rx", .stream_name = "HiFi Capture", .codec_dai_name = "sgtl5000", .ops = &mxs_sgtl5000_hifi_ops, + .capture_only = true, }, }; -- cgit v0.10.2 From bae35992afc7ca2f446108b16670d76de0624e7a Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 3 Sep 2013 15:14:12 -0300 Subject: ASoC: fsl: imx-audmux: Do not call imx_audmux_parse_dt_defaults() on non-dt kernel Booting a mx51babbage board with a non-dt kernel leads to the following crash: Unable to handle kernel NULL pointer dereference at virtual address 0000001c pgd = 80004000 [0000001c] *pgd=00000000 Internal error: Oops: 5 [#1] SMP ARM Modules linked in: CPU: 0 PID: 1 Comm: swapper/0 Not tainted 3.11.0-next-20130903 #287 task: 9f860000 ti: 9f862000 task.ti: 9f862000 PC is at of_get_next_available_child+0x5c/0x68 LR is at of_get_next_available_child+0x1c/0x68 pc : [<8043ea58>] lr : [<8043ea18>] psr: 60000193 sp : 9f863d58 ip : 00000000 fp : 9f863d74 r10: 9f89a010 r9 : 9f862000 r8 : 807bb26c r7 : 80615d5c r6 : 00000000 r5 : 60000113 r4 : 00000000 r3 : 00000000 r2 : 808770a4 r1 : 00000011 r0 : 60000113 Flags: nZCv IRQs off FIQs on Mode SVC_32 ISA ARM Segment kernel Control: 10c5387d Table: 90004019 DAC: 00000017 Process swapper/0 (pid: 1, stack limit = 0x9f862240) Stack: (0x9f863d58 to 0x9f864000) This is caused by commit 8548a464b9 (ASoC: imx-audmux: Read default configuration from devicetree). In order to fix this, add a check for 'of_id' so that imx_audmux_parse_dt_defaults() only gets called when a dt kernel is running. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ab17381..d3bf71a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev) if (audmux_type == IMX31_AUDMUX) audmux_debugfs_init(); - imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); + if (of_id) + imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node); return 0; } -- cgit v0.10.2 From 4345adf92db760ca1a54061ce284aaa2e7d0791e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 8 Sep 2013 17:20:37 +0100 Subject: ASoC: fsl_spdif: Select regmap-mmio The S/PDIF driver needs regmap so select it to make sure it gets included in the build. Reported-by: Fengguang Wu Acked-by: Nicolin Chen Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 704e246..b7ab71f 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -198,6 +198,7 @@ config SND_SOC_IMX_SPDIF select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF select SND_SOC_SPDIF + select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with -- cgit v0.10.2