From 16d11a829ed197b719723f81d82e7f1a42f5c681 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Jun 2009 14:07:53 +0200 Subject: ALSA: hda - Simplify AD1986A mixer definitions Split mixer element arrays of AD1986A models to several pieces so that each model can share the same mixer arrays. This removes lots of duplicated data. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 77 ++++++++++++-------------------------------- 1 file changed, 21 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 84cc49ca914..592423c878f 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -669,39 +669,13 @@ static struct hda_input_mux ad1986a_automic_capture_source = { }, }; -static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { +static 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), - HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, 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", 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", - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad198x_eapd_put, - .private_value = 0x1b | (1 << 8), /* port-D, inversed */ - }, { } /* end */ }; -static struct snd_kcontrol_new ad1986a_samsung_mixers[] = { - HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), - HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw), +static 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), @@ -727,6 +701,12 @@ static struct snd_kcontrol_new ad1986a_samsung_mixers[] = { { } /* end */ }; +static 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 */ +}; + /* re-connect the mic boost input according to the jack sensing */ static void ad1986a_automic(struct hda_codec *codec) { @@ -816,7 +796,7 @@ static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol, return change; } -static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = { +static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -826,33 +806,10 @@ static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = { .put = ad1986a_hp_master_sw_put, .private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT), }, - HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 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", 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", - .info = ad198x_eapd_info, - .get = ad198x_eapd_get, - .put = ad198x_eapd_put, - .private_value = 0x1b | (1 << 8), /* port-D, inversed */ - }, { } /* end */ }; + /* * initialization verbs */ @@ -1111,7 +1068,10 @@ static int patch_ad1986a(struct hda_codec *codec) spec->multiout.dac_nids = ad1986a_laptop_dac_nids; break; case AD1986A_LAPTOP_EAPD: - spec->mixers[0] = ad1986a_laptop_eapd_mixers; + 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; @@ -1122,7 +1082,9 @@ static int patch_ad1986a(struct hda_codec *codec) spec->input_mux = &ad1986a_laptop_eapd_capture_source; break; case AD1986A_SAMSUNG: - spec->mixers[0] = ad1986a_samsung_mixers; + 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; @@ -1136,7 +1098,10 @@ static int patch_ad1986a(struct hda_codec *codec) codec->patch_ops.init = ad1986a_automic_init; break; case AD1986A_LAPTOP_AUTOMUTE: - spec->mixers[0] = ad1986a_laptop_automute_mixers; + 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; -- cgit v1.2.3 From 03c405ad314d3c4e049b8d04500e54e833d16747 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Jun 2009 14:10:15 +0200 Subject: ALSA: hda - Generalize the pin-detect quirk for Lenovo N100 Add a new flag to ad_spec struct so that the same hack can be used for any other models (if any). This also allows other models to reuse the auto-mute functions. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 592423c878f..8c2b23f54f9 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -72,6 +72,7 @@ struct ad198x_spec { hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; unsigned int jack_present :1; + unsigned int inv_jack_detect:1; #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; @@ -756,8 +757,9 @@ static void ad1986a_hp_automute(struct hda_codec *codec) unsigned int present; present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0); - /* Lenovo N100 seems to report the reversed bit for HP jack-sensing */ - spec->jack_present = !(present & 0x80000000); + spec->jack_present = !!(present & 0x80000000); + if (spec->inv_jack_detect) + spec->jack_present = !spec->jack_present; ad1986a_update_hp(codec); } @@ -1113,6 +1115,10 @@ static int patch_ad1986a(struct hda_codec *codec) 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; case AD1986A_ULTRA: spec->mixers[0] = ad1986a_laptop_eapd_mixers; -- cgit v1.2.3 From c912e7a58054304575fe88574c776be7e684098e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 24 Jun 2009 14:14:34 +0200 Subject: ALSA: hda - Fix support for Samsung P50 with AD1986A codec Samsung P50 requires the HP auto-muting unlike other Samsung models. Added a new model=samsung-p50 to support this. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 41 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 8c2b23f54f9..1988582d1ab 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -940,6 +940,27 @@ static struct hda_verb ad1986a_hp_init_verbs[] = { {} }; +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 { @@ -950,6 +971,7 @@ enum { AD1986A_LAPTOP_AUTOMUTE, AD1986A_ULTRA, AD1986A_SAMSUNG, + AD1986A_SAMSUNG_P50, AD1986A_MODELS }; @@ -961,6 +983,7 @@ static const char *ad1986a_models[AD1986A_MODELS] = { [AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute", [AD1986A_ULTRA] = "ultra", [AD1986A_SAMSUNG] = "samsung", + [AD1986A_SAMSUNG_P50] = "samsung-p50", }; static struct snd_pci_quirk ad1986a_cfg_tbl[] = { @@ -983,6 +1006,7 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), 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), @@ -1099,6 +1123,23 @@ static int patch_ad1986a(struct hda_codec *codec) codec->patch_ops.unsol_event = ad1986a_automic_unsol_event; codec->patch_ops.init = ad1986a_automic_init; break; + case AD1986A_SAMSUNG_P50: + spec->num_mixers = 2; + spec->mixers[0] = ad1986a_automute_master_mixers; + spec->mixers[1] = ad1986a_laptop_eapd_mixers; + spec->num_init_verbs = 4; + spec->init_verbs[1] = ad1986a_eapd_init_verbs; + spec->init_verbs[2] = ad1986a_automic_verbs; + spec->init_verbs[3] = ad1986a_hp_init_verbs; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = ad1986a_laptop_dac_nids; + if (!is_jack_available(codec, 0x25)) + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1986a_automic_capture_source; + codec->patch_ops.unsol_event = ad1986a_samsung_p50_unsol_event; + codec->patch_ops.init = ad1986a_samsung_p50_init; + break; case AD1986A_LAPTOP_AUTOMUTE: spec->num_mixers = 3; spec->mixers[0] = ad1986a_automute_master_mixers; -- cgit v1.2.3 From 261c2407401ca26fa17f05667ea68f51e12c5303 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 25 Jun 2009 08:13:40 +0200 Subject: ALSA: hda - Add pin-sense trigger when needed for Realtek codecs Realtek codecs require the pin-sense trigger call before actually reading the pin-sense. Without this, the pin-detection might not be done accurately. This patch adds the pin-capability check and issues the trigger call if required. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33453319742..98ac24adf39 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -945,12 +945,13 @@ static void alc_fix_pll_init(struct hda_codec *codec, hda_nid_t nid, static void alc_automute_pin(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int present; + unsigned int present, pincap; unsigned int nid = spec->autocfg.hp_pins[0]; int i; - /* need to execute and sync at first */ - snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); + pincap = snd_hda_query_pin_caps(codec, nid); + if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ + snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0); present = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); spec->jack_present = (present & AC_PINSENSE_PRESENCE) != 0; @@ -1392,7 +1393,7 @@ static struct hda_verb alc888_fujitsu_xa3530_verbs[] = { static void alc_automute_amp(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - unsigned int val, mute; + unsigned int val, mute, pincap; hda_nid_t nid; int i; @@ -1401,6 +1402,10 @@ static void alc_automute_amp(struct hda_codec *codec) nid = spec->autocfg.hp_pins[i]; if (!nid) break; + pincap = snd_hda_query_pin_caps(codec, nid); + if (pincap & AC_PINCAP_TRIG_REQ) /* need trigger? */ + snd_hda_codec_read(codec, nid, 0, + AC_VERB_SET_PIN_SENSE, 0); val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_SENSE, 0); if (val & AC_PINSENSE_PRESENCE) { -- cgit v1.2.3 From 320d592001acbfd76bf856b5370319f144285489 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Emilio=20L=C3=B3pez?= Date: Thu, 25 Jun 2009 08:18:44 +0200 Subject: ALSA: hda - Fix acer-aspire-6530g model quirk MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix the following bugs of acer-aspire-6530g model with ALC888: - HP jack to mute all speaker outputs including LFE - Make digital built-in mic working Signed-off-by: Emilio López Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 98ac24adf39..7ebe5216b4b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1476,6 +1476,10 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { /* Bias voltage on for external mic port */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, +/* Front Mic: set to PIN_IN (empty by default) */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, +/* Unselect Front Mic by default in input mixer 3 */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, /* Enable unsolicited event for HP jack */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, /* Enable speaker output */ @@ -1644,6 +1648,17 @@ static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec) alc_automute_amp(codec); } +static void alc888_acer_aspire_6530g_init_hook(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + spec->autocfg.hp_pins[0] = 0x15; + spec->autocfg.speaker_pins[0] = 0x14; + spec->autocfg.speaker_pins[1] = 0x16; + spec->autocfg.speaker_pins[2] = 0x17; + alc_automute_amp(codec); +} + static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -9322,7 +9337,7 @@ static struct alc_config_preset alc883_presets[] = { ARRAY_SIZE(alc888_2_capture_sources), .input_mux = alc888_acer_aspire_6530_sources, .unsol_event = alc_automute_amp_unsol_event, - .init_hook = alc888_acer_aspire_4930g_init_hook, + .init_hook = alc888_acer_aspire_6530g_init_hook, }, [ALC888_ACER_ASPIRE_8930G] = { .mixers = { alc888_base_mixer, -- cgit v1.2.3 From dde6535686aa4e78e8b85850d1f3fccd8a581622 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 25 Jun 2009 08:25:35 +0200 Subject: ALSA: hda - Use model=acer-aspire-6530g for Acer Aspire 6930G For Acer Aspire 6930G (1025:015e), acre-aspire-6530g model matches obviously better. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7ebe5216b4b..2ed514030e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9084,7 +9084,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", - ALC888_ACER_ASPIRE_4930G), + ALC888_ACER_ASPIRE_6530G), SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", ALC888_ACER_ASPIRE_6530G), /* default Acer -- disabled as it causes more problems. -- cgit v1.2.3 From 28d27aae9432c300857722a917be4065c6d7abff Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Wed, 24 Jun 2009 22:13:35 -0700 Subject: sound: Use PCI_VDEVICE Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/oss/kahlua.c | 2 +- sound/pci/atiixp.c | 8 ++++---- sound/pci/atiixp_modem.c | 4 ++-- sound/pci/au88x0/au8810.c | 3 +-- sound/pci/au88x0/au8820.c | 3 +-- sound/pci/au88x0/au8830.c | 3 +-- sound/pci/cmipci.c | 10 +++++----- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx.c | 6 +++--- sound/pci/ens1370.c | 6 +++--- sound/pci/es1938.c | 2 +- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 46 ++++++++++++++++++++++----------------------- sound/pci/intel8x0m.c | 34 ++++++++++++++++----------------- sound/pci/mixart/mixart.c | 2 +- sound/pci/nm256/nm256.c | 6 +++--- sound/pci/rme32.c | 9 +++------ sound/pci/rme96.c | 12 ++++-------- sound/pci/sonicvibes.c | 2 +- sound/pci/via82xx.c | 4 ++-- sound/pci/via82xx_modem.c | 2 +- sound/pci/ymfpci/ymfpci.c | 12 ++++++------ 23 files changed, 86 insertions(+), 96 deletions(-) (limited to 'sound') diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index c180598f171..89466b056be 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -199,7 +199,7 @@ MODULE_LICENSE("GPL"); */ static struct pci_device_id id_tbl[] = { - { PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, + { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, { } }; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 71515ddb459..d6752dff2a4 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -287,10 +287,10 @@ struct atiixp { /* */ static struct pci_device_id snd_atiixp_ids[] = { - { 0x1002, 0x4341, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */ - { 0x1002, 0x4361, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB300 */ - { 0x1002, 0x4370, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */ - { 0x1002, 0x4382, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB600 */ + { PCI_VDEVICE(ATI, 0x4341), 0 }, /* SB200 */ + { PCI_VDEVICE(ATI, 0x4361), 0 }, /* SB300 */ + { PCI_VDEVICE(ATI, 0x4370), 0 }, /* SB400 */ + { PCI_VDEVICE(ATI, 0x4382), 0 }, /* SB600 */ { 0, } }; diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index c3136cccc55..e7e147bf8eb 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -262,8 +262,8 @@ struct atiixp_modem { /* */ static struct pci_device_id snd_atiixp_ids[] = { - { 0x1002, 0x434d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB200 */ - { 0x1002, 0x4378, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* SB400 */ + { PCI_VDEVICE(ATI, 0x434d), 0 }, /* SB200 */ + { PCI_VDEVICE(ATI, 0x4378), 0 }, /* SB400 */ { 0, } }; diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c index fce22c7af0e..c0e8c6b295c 100644 --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,8 +1,7 @@ #include "au8810.h" #include "au88x0.h" static struct pci_device_id snd_vortex_ids[] = { - {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1,}, + {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE), 1,}, {0,} }; diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c index d1fbcce0725..a6527330df5 100644 --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,8 +1,7 @@ #include "au8820.h" #include "au88x0.h" static struct pci_device_id snd_vortex_ids[] = { - {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, + {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1), 0,}, {0,} }; diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c index d4f2717c14f..6c702ad4352 100644 --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,8 +1,7 @@ #include "au8830.h" #include "au88x0.h" static struct pci_device_id snd_vortex_ids[] = { - {PCI_VENDOR_ID_AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, + {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2), 0,}, {0,} }; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 449fe02f666..ddcd4a9fd7e 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2797,11 +2797,11 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} static struct pci_device_id snd_cmipci_ids[] = { - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738B, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_AL, PCI_DEVICE_ID_CMEDIA_CM8738, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A), 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B), 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, + {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738B), 0}, + {PCI_VDEVICE(AL, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, {0,}, }; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index f6286f84a22..e2e0359bb05 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -495,7 +495,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); static struct pci_device_id snd_cs4281_ids[] = { - { 0x1013, 0x6005, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4281 */ + { PCI_VDEVICE(CIRRUS, 0x6005), 0, }, /* CS4281 */ { 0, } }; diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index c9b3e3d48cb..033aec43011 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -65,9 +65,9 @@ module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); static struct pci_device_id snd_cs46xx_ids[] = { - { 0x1013, 0x6001, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4280 */ - { 0x1013, 0x6003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4612 */ - { 0x1013, 0x6004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* CS4615 */ + { PCI_VDEVICE(CIRRUS, 0x6001), 0, }, /* CS4280 */ + { PCI_VDEVICE(CIRRUS, 0x6003), 0, }, /* CS4612 */ + { PCI_VDEVICE(CIRRUS, 0x6004), 0, }, /* CS4615 */ { 0, } }; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 18f4d1e98c4..d589bbc516e 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -445,11 +445,11 @@ static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id); static struct pci_device_id snd_audiopci_ids[] = { #ifdef CHIP1370 - { 0x1274, 0x5000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1370 */ + { PCI_VDEVICE(ENSONIQ, 0x5000), 0, }, /* ES1370 */ #endif #ifdef CHIP1371 - { 0x1274, 0x1371, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1371 */ - { 0x1274, 0x5880, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ES1373 - CT5880 */ + { PCI_VDEVICE(ENSONIQ, 0x1371), 0, }, /* ES1371 */ + { PCI_VDEVICE(ENSONIQ, 0x5880), 0, }, /* ES1373 - CT5880 */ { 0x1102, 0x8938, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Ectiva EV1938 */ #endif { 0, } diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fbd2ac09aa3..820318ee62c 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -244,7 +244,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); static struct pci_device_id snd_es1938_ids[] = { - { 0x125d, 0x1969, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Solo-1 */ + { PCI_VDEVICE(ESS, 0x1969), 0, }, /* Solo-1 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 0d0cdbdb448..cecf1ffeeaa 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -107,7 +107,7 @@ MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); static const struct pci_device_id snd_ice1712_ids[] = { - { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_ICE_1712, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICE1712 */ + { PCI_VDEVICE(ICE, PCI_DEVICE_ID_ICE_1712), 0 }, /* ICE1712 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 36ade77cf37..cc84a831eb2 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -93,7 +93,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ static const struct pci_device_id snd_vt1724_ids[] = { - { PCI_VENDOR_ID_ICE, PCI_DEVICE_ID_VT1724, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, + { PCI_VDEVICE(ICE, PCI_DEVICE_ID_VT1724), 0 }, { 0, } }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 8aa5687f392..171ada53520 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -421,29 +421,29 @@ struct intel8x0 { }; static struct pci_device_id snd_intel8x0_ids[] = { - { 0x8086, 0x2415, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */ - { 0x8086, 0x2425, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */ - { 0x8086, 0x2445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */ - { 0x8086, 0x2485, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH3 */ - { 0x8086, 0x24c5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH4 */ - { 0x8086, 0x24d5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH5 */ - { 0x8086, 0x25a6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ESB */ - { 0x8086, 0x266e, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH6 */ - { 0x8086, 0x27de, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ICH7 */ - { 0x8086, 0x2698, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL_ICH4 }, /* ESB2 */ - { 0x8086, 0x7195, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 440MX */ - { 0x1039, 0x7012, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_SIS }, /* SI7012 */ - { 0x10de, 0x01b1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE */ - { 0x10de, 0x003a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* MCP04 */ - { 0x10de, 0x006a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2 */ - { 0x10de, 0x0059, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK804 */ - { 0x10de, 0x008a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK8 */ - { 0x10de, 0x00da, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE3 */ - { 0x10de, 0x00ea, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* CK8S */ - { 0x10de, 0x026b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* MCP51 */ - { 0x1022, 0x746d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD8111 */ - { 0x1022, 0x7445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD768 */ - { 0x10b9, 0x5455, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALI }, /* Ali5455 */ + { PCI_VDEVICE(INTEL, 0x2415), DEVICE_INTEL }, /* 82801AA */ + { PCI_VDEVICE(INTEL, 0x2425), DEVICE_INTEL }, /* 82901AB */ + { PCI_VDEVICE(INTEL, 0x2445), DEVICE_INTEL }, /* 82801BA */ + { PCI_VDEVICE(INTEL, 0x2485), DEVICE_INTEL }, /* ICH3 */ + { PCI_VDEVICE(INTEL, 0x24c5), DEVICE_INTEL_ICH4 }, /* ICH4 */ + { PCI_VDEVICE(INTEL, 0x24d5), DEVICE_INTEL_ICH4 }, /* ICH5 */ + { PCI_VDEVICE(INTEL, 0x25a6), DEVICE_INTEL_ICH4 }, /* ESB */ + { PCI_VDEVICE(INTEL, 0x266e), DEVICE_INTEL_ICH4 }, /* ICH6 */ + { PCI_VDEVICE(INTEL, 0x27de), DEVICE_INTEL_ICH4 }, /* ICH7 */ + { PCI_VDEVICE(INTEL, 0x2698), DEVICE_INTEL_ICH4 }, /* ESB2 */ + { PCI_VDEVICE(INTEL, 0x7195), DEVICE_INTEL }, /* 440MX */ + { PCI_VDEVICE(SI, 0x7012), DEVICE_SIS }, /* SI7012 */ + { PCI_VDEVICE(NVIDIA, 0x01b1), DEVICE_NFORCE }, /* NFORCE */ + { PCI_VDEVICE(NVIDIA, 0x003a), DEVICE_NFORCE }, /* MCP04 */ + { PCI_VDEVICE(NVIDIA, 0x006a), DEVICE_NFORCE }, /* NFORCE2 */ + { PCI_VDEVICE(NVIDIA, 0x0059), DEVICE_NFORCE }, /* CK804 */ + { PCI_VDEVICE(NVIDIA, 0x008a), DEVICE_NFORCE }, /* CK8 */ + { PCI_VDEVICE(NVIDIA, 0x00da), DEVICE_NFORCE }, /* NFORCE3 */ + { PCI_VDEVICE(NVIDIA, 0x00ea), DEVICE_NFORCE }, /* CK8S */ + { PCI_VDEVICE(NVIDIA, 0x026b), DEVICE_NFORCE }, /* MCP51 */ + { PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL }, /* AMD8111 */ + { PCI_VDEVICE(AMD, 0x7445), DEVICE_INTEL }, /* AMD768 */ + { PCI_VDEVICE(AL, 0x5455), DEVICE_ALI }, /* Ali5455 */ { 0, } }; diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 6ec0fc50d6b..9e7d12e7673 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -220,24 +220,24 @@ struct intel8x0m { }; static struct pci_device_id snd_intel8x0m_ids[] = { - { 0x8086, 0x2416, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801AA */ - { 0x8086, 0x2426, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82901AB */ - { 0x8086, 0x2446, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 82801BA */ - { 0x8086, 0x2486, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH3 */ - { 0x8086, 0x24c6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH4 */ - { 0x8086, 0x24d6, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH5 */ - { 0x8086, 0x266d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH6 */ - { 0x8086, 0x27dd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* ICH7 */ - { 0x8086, 0x7196, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* 440MX */ - { 0x1022, 0x7446, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD768 */ - { 0x1039, 0x7013, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_SIS }, /* SI7013 */ - { 0x10de, 0x01c1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE */ - { 0x10de, 0x0069, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2 */ - { 0x10de, 0x0089, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE2s */ - { 0x10de, 0x00d9, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_NFORCE }, /* NFORCE3 */ + { PCI_VDEVICE(INTEL, 0x2416), DEVICE_INTEL }, /* 82801AA */ + { PCI_VDEVICE(INTEL, 0x2426), DEVICE_INTEL }, /* 82901AB */ + { PCI_VDEVICE(INTEL, 0x2446), DEVICE_INTEL }, /* 82801BA */ + { PCI_VDEVICE(INTEL, 0x2486), DEVICE_INTEL }, /* ICH3 */ + { PCI_VDEVICE(INTEL, 0x24c6), DEVICE_INTEL }, /* ICH4 */ + { PCI_VDEVICE(INTEL, 0x24d6), DEVICE_INTEL }, /* ICH5 */ + { PCI_VDEVICE(INTEL, 0x266d), DEVICE_INTEL }, /* ICH6 */ + { PCI_VDEVICE(INTEL, 0x27dd), DEVICE_INTEL }, /* ICH7 */ + { PCI_VDEVICE(INTEL, 0x7196), DEVICE_INTEL }, /* 440MX */ + { PCI_VDEVICE(AMD, 0x7446), DEVICE_INTEL }, /* AMD768 */ + { PCI_VDEVICE(SI, 0x7013), DEVICE_SIS }, /* SI7013 */ + { PCI_VDEVICE(NVIDIA, 0x01c1), DEVICE_NFORCE }, /* NFORCE */ + { PCI_VDEVICE(NVIDIA, 0x0069), DEVICE_NFORCE }, /* NFORCE2 */ + { PCI_VDEVICE(NVIDIA, 0x0089), DEVICE_NFORCE }, /* NFORCE2s */ + { PCI_VDEVICE(NVIDIA, 0x00d9), DEVICE_NFORCE }, /* NFORCE3 */ #if 0 - { 0x1022, 0x746d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL }, /* AMD8111 */ - { 0x10b9, 0x5455, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALI }, /* Ali5455 */ + { PCI_VDEVICE(AMD, 0x746d), DEVICE_INTEL }, /* AMD8111 */ + { PCI_VDEVICE(AL, 0x5455), DEVICE_ALI }, /* Ali5455 */ #endif { 0, } }; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 82bc5b9e762..a83d1968a84 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -61,7 +61,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); */ static struct pci_device_id snd_mixart_ids[] = { - { 0x1057, 0x0003, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* MC8240 */ + { PCI_VDEVICE(MOTOROLA, 0x0003), 0, }, /* MC8240 */ { 0, } }; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 522a040855d..97a0731331a 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -263,9 +263,9 @@ struct nm256 { * PCI ids */ static struct pci_device_id snd_nm256_ids[] = { - {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, - {PCI_VENDOR_ID_NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0}, + {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO), 0}, + {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO), 0}, + {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO), 0}, {0,}, }; diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index d7b966e7c4c..f977dba7cbd 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -227,12 +227,9 @@ struct rme32 { }; static struct pci_device_id snd_rme32_ids[] = { - {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, - {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, - {PCI_VENDOR_ID_XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0,}, + {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32), 0,}, + {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8), 0,}, + {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO), 0,}, {0,} }; diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 55fb1c131f5..2ba5c0fd55d 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -232,14 +232,10 @@ struct rme96 { }; static struct pci_device_id snd_rme96_ids[] = { - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, - { PCI_VENDOR_ID_XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96), 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8), 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO), 0, }, + { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PAD_OR_PST), 0, }, { 0, } }; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 7dc60ad4772..1f6406c4534 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -243,7 +243,7 @@ struct sonicvibes { }; static struct pci_device_id snd_sonic_ids[] = { - { 0x5333, 0xca00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, + { PCI_VDEVICE(S3, 0xca00), 0, }, { 0, } }; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 949fcaf6b70..acfa4760da4 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -402,9 +402,9 @@ struct via82xx { static struct pci_device_id snd_via82xx_ids[] = { /* 0x1106, 0x3058 */ - { PCI_VENDOR_ID_VIA, PCI_DEVICE_ID_VIA_82C686_5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA686, }, /* 686A */ + { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_82C686_5), TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ - { PCI_VENDOR_ID_VIA, PCI_DEVICE_ID_VIA_8233_5, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA8233, }, /* VT8233 */ + { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_8233_5), TYPE_CARD_VIA8233, }, /* VT8233 */ { 0, } }; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 0d54e3503c1..47eb61561df 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -261,7 +261,7 @@ struct via82xx_modem { }; static struct pci_device_id snd_via82xx_modem_ids[] = { - { 0x1106, 0x3068, PCI_ANY_ID, PCI_ANY_ID, 0, 0, TYPE_CARD_VIA82XX_MODEM, }, + { PCI_VDEVICE(VIA, 0x3068), TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 4af66661f9b..e6b18b90d45 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -67,12 +67,12 @@ module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); static struct pci_device_id snd_ymfpci_ids[] = { - { 0x1073, 0x0004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724 */ - { 0x1073, 0x000d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF724F */ - { 0x1073, 0x000a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF740 */ - { 0x1073, 0x000c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF740C */ - { 0x1073, 0x0010, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF744 */ - { 0x1073, 0x0012, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* YMF754 */ + { PCI_VDEVICE(YAMAHA, 0x0004), 0, }, /* YMF724 */ + { PCI_VDEVICE(YAMAHA, 0x000d), 0, }, /* YMF724F */ + { PCI_VDEVICE(YAMAHA, 0x000a), 0, }, /* YMF740 */ + { PCI_VDEVICE(YAMAHA, 0x000c), 0, }, /* YMF740C */ + { PCI_VDEVICE(YAMAHA, 0x0010), 0, }, /* YMF744 */ + { PCI_VDEVICE(YAMAHA, 0x0012), 0, }, /* YMF754 */ { 0, } }; -- cgit v1.2.3 From 0d7392e54435476243ce08ba57745ab52d639cbb Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Wed, 24 Jun 2009 23:18:02 -0700 Subject: sound: Use PCI_VDEVICE for CREATIVE and ECTIVA Here's a patch on top of the others to use CREATIVE and ECTIVA Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/emu10k1/emu10k1.c | 6 +++--- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/ens1370.c | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 57b992a5c05..f24bf1ecb36 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1876,7 +1876,7 @@ static int snd_ca0106_resume(struct pci_dev *pci) // PCI IDs static struct pci_device_id snd_ca0106_ids[] = { - { 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Audigy LS or Live 24bit */ + { PCI_VDEVICE(CREATIVE, 0x0007), 0 }, /* Audigy LS or Live 24bit */ { 0, } }; MODULE_DEVICE_TABLE(pci, snd_ca0106_ids); diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index c7f3b994101..168af67d938 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -77,9 +77,9 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ static struct pci_device_id snd_emu10k1_ids[] = { - { 0x1102, 0x0002, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* EMU10K1 */ - { 0x1102, 0x0004, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1 }, /* Audigy */ - { 0x1102, 0x0008, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 1 }, /* Audigy 2 Value SB0400 */ + { PCI_VDEVICE(CREATIVE, 0x0002), 0 }, /* EMU10K1 */ + { PCI_VDEVICE(CREATIVE, 0x0004), 1 }, /* Audigy */ + { PCI_VDEVICE(CREATIVE, 0x0008), 1 }, /* Audigy 2 Value SB0400 */ { 0, } }; diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 4d3ad793e98..36e08bd2b3c 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1607,7 +1607,7 @@ static void __devexit snd_emu10k1x_remove(struct pci_dev *pci) // PCI IDs static struct pci_device_id snd_emu10k1x_ids[] = { - { 0x1102, 0x0006, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Dell OEM version (EMU10K1) */ + { PCI_VDEVICE(CREATIVE, 0x0006), 0 }, /* Dell OEM version (EMU10K1) */ { 0, } }; MODULE_DEVICE_TABLE(pci, snd_emu10k1x_ids); diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index d589bbc516e..2b82c5c723e 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -450,7 +450,7 @@ static struct pci_device_id snd_audiopci_ids[] = { #ifdef CHIP1371 { PCI_VDEVICE(ENSONIQ, 0x1371), 0, }, /* ES1371 */ { PCI_VDEVICE(ENSONIQ, 0x5880), 0, }, /* ES1373 - CT5880 */ - { 0x1102, 0x8938, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* Ectiva EV1938 */ + { PCI_VDEVICE(ECTIVA, 0x8938), 0, }, /* Ectiva EV1938 */ #endif { 0, } }; -- cgit v1.2.3 From 7e895cfaad51c862932ea7db0c428761076412e5 Mon Sep 17 00:00:00 2001 From: Tim Blechmann Date: Thu, 25 Jun 2009 09:41:46 +0200 Subject: ALSA: lx6464es - configure ethersound io channels as long as the io channel number is not set by the driver, the card is not visible from the ethersound network Signed-off-by: Tim Blechmann Signed-off-by: Takashi Iwai --- sound/pci/lx6464es/lx6464es.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 18da2ef04d0..11b8c6514b3 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -654,13 +654,12 @@ static int __devinit lx_init_ethersound_config(struct lx6464es *chip) int i; u32 orig_conf_es = lx_dsp_reg_read(chip, eReg_CONFES); - u32 default_conf_es = (64 << IOCR_OUTPUTS_OFFSET) | + /* configure 64 io channels */ + u32 conf_es = (orig_conf_es & CONFES_READ_PART_MASK) | (64 << IOCR_INPUTS_OFFSET) | + (64 << IOCR_OUTPUTS_OFFSET) | (FREQ_RATIO_SINGLE_MODE << FREQ_RATIO_OFFSET); - u32 conf_es = (orig_conf_es & CONFES_READ_PART_MASK) - | (default_conf_es & CONFES_WRITE_PART_MASK); - snd_printdd("->lx_init_ethersound\n"); chip->freq_ratio = FREQ_RATIO_SINGLE_MODE; -- cgit v1.2.3 From 14744d7da2e6ab5c6d8e82c84dc280e3c0dd8552 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 25 Jun 2009 14:28:49 +0200 Subject: sound: oxygen: make mic volume control mono The microphone input and its volume register have only one channel, so we have to make the corresponding mixer control a mono control. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_mixer.c | 28 ++++++++++++++++++---------- 1 file changed, 18 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 304da169bfd..5401c547c4e 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -575,8 +575,10 @@ static int ac97_switch_put(struct snd_kcontrol *ctl, static int ac97_volume_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) { + int stereo = (ctl->private_value >> 16) & 1; + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - info->count = 2; + info->count = stereo ? 2 : 1; info->value.integer.min = 0; info->value.integer.max = 0x1f; return 0; @@ -587,6 +589,7 @@ static int ac97_volume_get(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; unsigned int codec = (ctl->private_value >> 24) & 1; + int stereo = (ctl->private_value >> 16) & 1; unsigned int index = ctl->private_value & 0xff; u16 reg; @@ -594,7 +597,8 @@ static int ac97_volume_get(struct snd_kcontrol *ctl, reg = oxygen_read_ac97(chip, codec, index); mutex_unlock(&chip->mutex); value->value.integer.value[0] = 31 - (reg & 0x1f); - value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f); + if (stereo) + value->value.integer.value[1] = 31 - ((reg >> 8) & 0x1f); return 0; } @@ -603,6 +607,7 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; unsigned int codec = (ctl->private_value >> 24) & 1; + int stereo = (ctl->private_value >> 16) & 1; unsigned int index = ctl->private_value & 0xff; u16 oldreg, newreg; int change; @@ -612,8 +617,11 @@ static int ac97_volume_put(struct snd_kcontrol *ctl, newreg = oldreg; newreg = (newreg & ~0x1f) | (31 - (value->value.integer.value[0] & 0x1f)); - newreg = (newreg & ~0x1f00) | - ((31 - (value->value.integer.value[0] & 0x1f)) << 8); + if (stereo) + newreg = (newreg & ~0x1f00) | + ((31 - (value->value.integer.value[1] & 0x1f)) << 8); + else + newreg = (newreg & ~0x1f00) | ((newreg & 0x1f) << 8); change = newreg != oldreg; if (change) oxygen_write_ac97(chip, codec, index, newreg); @@ -673,7 +681,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, .private_value = ((codec) << 24) | ((invert) << 16) | \ ((bitnr) << 8) | (index), \ } -#define AC97_VOLUME(xname, codec, index) { \ +#define AC97_VOLUME(xname, codec, index, stereo) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ @@ -682,7 +690,7 @@ static int ac97_fp_rec_volume_put(struct snd_kcontrol *ctl, .get = ac97_volume_get, \ .put = ac97_volume_put, \ .tlv = { .p = ac97_db_scale, }, \ - .private_value = ((codec) << 24) | (index), \ + .private_value = ((codec) << 24) | ((stereo) << 16) | (index), \ } static DECLARE_TLV_DB_SCALE(monitor_db_scale, -1000, 1000, 0); @@ -882,18 +890,18 @@ static const struct { }; static const struct snd_kcontrol_new ac97_controls[] = { - AC97_VOLUME("Mic Capture Volume", 0, AC97_MIC), + AC97_VOLUME("Mic Capture Volume", 0, AC97_MIC, 0), AC97_SWITCH("Mic Capture Switch", 0, AC97_MIC, 15, 1), AC97_SWITCH("Mic Boost (+20dB)", 0, AC97_MIC, 6, 0), AC97_SWITCH("Line Capture Switch", 0, AC97_LINE, 15, 1), - AC97_VOLUME("CD Capture Volume", 0, AC97_CD), + AC97_VOLUME("CD Capture Volume", 0, AC97_CD, 1), AC97_SWITCH("CD Capture Switch", 0, AC97_CD, 15, 1), - AC97_VOLUME("Aux Capture Volume", 0, AC97_AUX), + AC97_VOLUME("Aux Capture Volume", 0, AC97_AUX, 1), AC97_SWITCH("Aux Capture Switch", 0, AC97_AUX, 15, 1), }; static const struct snd_kcontrol_new ac97_fp_controls[] = { - AC97_VOLUME("Front Panel Playback Volume", 1, AC97_HEADPHONE), + AC97_VOLUME("Front Panel Playback Volume", 1, AC97_HEADPHONE, 1), AC97_SWITCH("Front Panel Playback Switch", 1, AC97_HEADPHONE, 15, 1), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, -- cgit v1.2.3 From 684a88429c5ab04d8b1894de9a1ef62de6f601b7 Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Fri, 26 Jun 2009 09:27:50 +0100 Subject: ALSA: hda - Line In for Acer Inspire 6530G model The Line In connector is set up as PIN_IN by default, using VREF_HIZ. It is connected to both ADCs, so add it to both input selectors. Also add the ability to use the input mix (on a SoundBlaster one would call this "What You Hear"). Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2ed514030e7..08846d222cb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1569,18 +1569,22 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { /* Interal mic only available on one ADC */ { - .num_items = 3, + .num_items = 5, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, { "Int Mic", 0xb }, }, }, { - .num_items = 2, + .num_items = 4, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, }, } }; @@ -8209,6 +8213,8 @@ static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), -- cgit v1.2.3 From c2a30d711852e4f39c8a79135b3caa701f7a8e02 Mon Sep 17 00:00:00 2001 From: Ondrej Zary Date: Sat, 27 Jun 2009 16:17:08 +0200 Subject: ALSA: cmi8330: fix MPU-401 PnP init copy&paste bug Fix copy&paste bug in PnP MPU-401 initialization. Signed-off-by: Ondrej Zary Cc: Signed-off-by: Takashi Iwai --- sound/isa/cmi8330.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index de83608719e..3ee0269e5bd 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -338,7 +338,7 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, return -EBUSY; acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); - if (acard->play == NULL) + if (acard->mpu == NULL) return -EBUSY; pdev = acard->cap; -- cgit v1.2.3 From 1d955ebd4db7961dc4e772a23288b3d5c6f191be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Jun 2009 11:33:53 +0200 Subject: ALSA: hda - Add missing initializations for ALC268 and ALC269 During the changes to clean up / fix the realtek codec initialization routines in commit 4a79ba34cada6a5a4ee86ed53aa8a73ba1e6fc51, I forgot to add the check for ALC268 and ALC269. This resulted in the missing EAPD and COEF setup for these codecs. This patch adds the missing checks for these codecs. Reference: bko#13633 http://bugzilla.kernel.org/show_bug.cgi?id=13633 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 08846d222cb..3a8e58c483d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12463,6 +12463,8 @@ static int alc268_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } @@ -13371,6 +13373,8 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); + alc_ssid_check(codec, 0x15, 0x1b, 0x14); + return 1; } -- cgit v1.2.3 From 9ea5ca75a2aebb7172094a7d77acf6ff7600cc56 Mon Sep 17 00:00:00 2001 From: Jiri Slaby Date: Mon, 29 Jun 2009 18:03:34 +0200 Subject: sound: OSS: mpu401, fix deadlock mpu401_chk_version is called with a spin lock already held. Don't take it again. Signed-off-by: Jiri Slaby Signed-off-by: Takashi Iwai --- sound/oss/mpu401.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) (limited to 'sound') diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 6c0a770ed05..1b2316f35b1 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -926,31 +926,21 @@ static struct midi_operations mpu401_midi_operations[MAX_MIDI_DEV]; static void mpu401_chk_version(int n, struct mpu_config *devc) { int tmp; - unsigned long flags; devc->version = devc->revision = 0; - spin_lock_irqsave(&devc->lock,flags); - if ((tmp = mpu_cmd(n, 0xAC, 0)) < 0) - { - spin_unlock_irqrestore(&devc->lock,flags); + tmp = mpu_cmd(n, 0xAC, 0); + if (tmp < 0) return; - } if ((tmp & 0xf0) > 0x20) /* Why it's larger than 2.x ??? */ - { - spin_unlock_irqrestore(&devc->lock,flags); return; - } devc->version = tmp; - if ((tmp = mpu_cmd(n, 0xAD, 0)) < 0) - { + if ((tmp = mpu_cmd(n, 0xAD, 0)) < 0) { devc->version = 0; - spin_unlock_irqrestore(&devc->lock,flags); return; } devc->revision = tmp; - spin_unlock_irqrestore(&devc->lock,flags); } int attach_mpu401(struct address_info *hw_config, struct module *owner) -- cgit v1.2.3 From 6a84c234da06a4ac0c1b4c819b83cf264674c2d8 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Sun, 28 Jun 2009 01:41:52 -0600 Subject: ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfig ALSA SoC drivers should be specify SND_SOC_AC97_BUS instead, not AC97_BUS. Without SND_SOC_AC97_BUS defined, an AC97 device will not get correctly registered on the AC97 bus, which prevents thinks like the WM9712 touchscreen driver from getting probed. Tested against 2.6.31-rc1. Signed-off-by: Grant Likely Acked-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 5dbebf82249..5661876ee83 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -33,7 +33,7 @@ config SND_SOC_MPC5200_I2S config SND_SOC_MPC5200_AC97 tristate "Freescale MPC5200 PSC in AC97 mode driver" depends on PPC_MPC52xx && PPC_BESTCOMM - select AC97_BUS + select SND_SOC_AC97_BUS select SND_MPC52xx_DMA select PPC_BESTCOMM_GEN_BD help -- cgit v1.2.3 From 40d9ec14e7e1f62d2379ecc1b5ee00ddfc2a5d0c Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Sun, 28 Jun 2009 01:42:06 -0600 Subject: ASoC: remove BROKEN from Efika and pcm030 fabric drivers The needed spin_event_timeout() macro is now merged in from the powerpc tree, so these drivers are no longer broken. This reverts commit 0c0e09e21a9e7bc6ca54e06ef3d497255ca26383 (ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved) Tested against 2.6.31-rc1. Signed-off-by: Grant Likely Acked-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 5661876ee83..8cb65ccad35 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -41,7 +41,7 @@ config SND_SOC_MPC5200_AC97 config SND_MPC52xx_SOC_PCM030 tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" - depends on PPC_MPC5200_SIMPLE && BROKEN + depends on PPC_MPC5200_SIMPLE select SND_SOC_MPC5200_AC97 select SND_SOC_WM9712 help @@ -50,7 +50,7 @@ config SND_MPC52xx_SOC_PCM030 config SND_MPC52xx_SOC_EFIKA tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" - depends on PPC_EFIKA && BROKEN + depends on PPC_EFIKA select SND_SOC_MPC5200_AC97 select SND_SOC_STAC9766 help -- cgit v1.2.3 From 1bdd7419910c1506151e7b9e2d60c6980e015f76 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Sun, 28 Jun 2009 00:21:05 +0200 Subject: ASoC: OMAP: fix OMAP1510 broken PCM pointer callback This patch tries to work around the problem of broken OMAP1510 PCM playback pointer calculation by replacing DMA function call that incorrectly tries to read the value form DMA hardware with a value computed locally from an already maintained variable omap_runtime_data.period_index. Tested on OMAP5910 based Amstrad Delta (E3) using work in progress ASoC driver. Based on linux-2.6-asoc.git v2.6.31-rc1. Signed-off-by: Janusz Krzysztofik Acked-by: Jarkko Nikula Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 6454e15f7d2..84a1950880e 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -216,12 +216,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - ptr = omap_get_dma_src_pos(prtd->dma_ch); - else + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else if (!(cpu_is_omap1510())) { + ptr = omap_get_dma_src_pos(prtd->dma_ch); + offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); + } else + offset = prtd->period_index * runtime->period_size; - offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); if (offset >= runtime->buffer_size) offset = 0; -- cgit v1.2.3 From 1e1689536f346a431b748dc8ad9ac0828d2c065d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Jul 2009 08:34:32 +0200 Subject: ALSA: hda - Add missing static to patch_ca0110() Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_ca0110.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c index 392d108c355..019ca7cb56d 100644 --- a/sound/pci/hda/patch_ca0110.c +++ b/sound/pci/hda/patch_ca0110.c @@ -510,7 +510,7 @@ static int ca0110_parse_auto_config(struct hda_codec *codec) } -int patch_ca0110(struct hda_codec *codec) +static int patch_ca0110(struct hda_codec *codec) { struct ca0110_spec *spec; int err; -- cgit v1.2.3 From ff84847171508a3c76eb7e483204d1be7738729b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 1 Jul 2009 18:08:01 +0200 Subject: ALSA: hda - Add quirk for HP 6930p Added a quirk model=laptop for HP 6930p (103c:30dc) with AD1984A codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 84cc49ca914..85e8618e849 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3966,6 +3966,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { 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(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), -- cgit v1.2.3 From da9ff1f796e81976935407251815838bef9868d4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 1 Jul 2009 18:23:26 +0100 Subject: ASoC: Only disable pxa2xx-i2s clocks if we enabled them The clock API can't cope with unbalanced enables and disables and we only enable in hw_params() but try to disable in shutdown. Signed-off-by: Mark Brown --- sound/soc/pxa/pxa2xx-i2s.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 4743e262895..6b8f655d1ad 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -167,6 +167,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); + dai->private_data = dai; pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -255,7 +256,10 @@ static void pxa2xx_i2s_shutdown(struct snd_pcm_substream *substream, if ((SACR1 & (SACR1_DREC | SACR1_DRPL)) == (SACR1_DREC | SACR1_DRPL)) { SACR0 &= ~SACR0_ENB; pxa_i2s_wait(); - clk_disable(clk_i2s); + if (dai->private_data != NULL) { + clk_disable(clk_i2s); + dai->private_data = NULL; + } } } @@ -336,6 +340,7 @@ static int pxa2xx_i2s_probe(struct platform_device *dev) return PTR_ERR(clk_i2s); pxa_i2s_dai.dev = &dev->dev; + pxa_i2s_dai.private_data = NULL; ret = snd_soc_register_dai(&pxa_i2s_dai); if (ret != 0) clk_put(clk_i2s); -- cgit v1.2.3 From 826390796d09444b93e1f957582f8970ddfd9b3d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 2 Jul 2009 08:31:30 +0200 Subject: sound: virtuoso: fix Xonar D1/DX silence after resume When resuming, we better take the DACs out of the reset state before trying to use them. Reference: kernel bug #13599 http://bugzilla.kernel.org/show_bug.cgi?id=13599 Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/virtuoso.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index bf971f7cfdc..6ebcb6bdd71 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -635,6 +635,8 @@ static void xonar_d2_resume(struct oxygen *chip) static void xonar_d1_resume(struct oxygen *chip) { + oxygen_set_bits8(chip, OXYGEN_FUNCTION, OXYGEN_FUNCTION_RESET_CODEC); + msleep(1); cs43xx_init(chip); xonar_enable_output(chip); } -- cgit v1.2.3 From 563c2bf59d392357bcc1d99642933cc88c687964 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 2 Jul 2009 10:46:35 +0200 Subject: ALSA: snd_usb_caiaq: reparent sound device The sound device instance needs to be a child of the USB interface, not the USB device. Newer udev versions pay attention to that. Signed-off-by: Daniel Mack Reported-by: Lennart Poettering Signed-off-by: Takashi Iwai --- sound/usb/caiaq/device.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 0e5db719de2..de38108f0b2 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack "); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.18"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," @@ -349,7 +349,9 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) log("Unable to set up control system (ret=%d)\n", ret); } -static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) +static int create_card(struct usb_device *usb_dev, + struct usb_interface *intf, + struct snd_card **cardp) { int devnum; int err; @@ -374,7 +376,7 @@ static int create_card(struct usb_device* usb_dev, struct snd_card **cardp) dev->chip.usb_id = USB_ID(le16_to_cpu(usb_dev->descriptor.idVendor), le16_to_cpu(usb_dev->descriptor.idProduct)); spin_lock_init(&dev->spinlock); - snd_card_set_dev(card, &usb_dev->dev); + snd_card_set_dev(card, &intf->dev); *cardp = card; return 0; @@ -461,7 +463,7 @@ static int __devinit snd_probe(struct usb_interface *intf, struct snd_card *card; struct usb_device *device = interface_to_usbdev(intf); - ret = create_card(device, &card); + ret = create_card(device, intf, &card); if (ret < 0) return ret; -- cgit v1.2.3 From 3f5d3465be8f6e04f43d9b6d543fe28d4be07d78 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Jul 2009 11:51:44 +0200 Subject: ALSA: usx2y - reparent sound device Fix the parent device to be the USB interface, not the USB device. A similiar commit like 563c2bf59d392357bcc1d99642933cc88c687964. Signed-off-by: Takashi Iwai --- sound/usb/usx2y/us122l.c | 2 +- sound/usb/usx2y/usbusx2y.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index a5aae9d67f3..fd44946ce4b 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -514,7 +514,6 @@ static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp) US122L(card)->chip.dev->bus->busnum, US122L(card)->chip.dev->devnum ); - snd_card_set_dev(card, &device->dev); *cardp = card; return 0; } @@ -531,6 +530,7 @@ static int us122l_usb_probe(struct usb_interface *intf, if (err < 0) return err; + snd_card_set_dev(card, &intf->dev); if (!us122l_create_card(card)) { snd_card_free(card); return -EINVAL; diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index 5ce0da23ee9..cb4bb8373ca 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -364,7 +364,6 @@ static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp) 0,//us428(card)->usbmidi.ifnum, usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum ); - snd_card_set_dev(card, &device->dev); *cardp = card; return 0; } @@ -388,6 +387,7 @@ static int usX2Y_usb_probe(struct usb_device *device, err = usX2Y_create_card(device, &card); if (err < 0) return err; + snd_card_set_dev(card, &intf->dev); if ((err = usX2Y_hwdep_new(card, device)) < 0 || (err = snd_card_register(card)) < 0) { snd_card_free(card); -- cgit v1.2.3 From 1df892cba45f9856d369a6a317ad2d1e44bca423 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jul 2009 10:33:39 +0100 Subject: ASoC: Fix register cache initialisation for WM8753 The wrong register cache variable was being used to provide the size for the memcpy(), resulting in a copy of only a void * of data. Reported-by: Lars-Peter Clausen Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8753.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index d28eeaceb85..e06b0cfe4f2 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1664,7 +1664,7 @@ static int wm8753_register(struct wm8753_priv *wm8753) codec->reg_cache = &wm8753->reg_cache; codec->private_data = wm8753; - memcpy(codec->reg_cache, wm8753_reg, sizeof(codec->reg_cache)); + memcpy(codec->reg_cache, wm8753_reg, sizeof(wm8753->reg_cache)); INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work); ret = wm8753_reset(codec); -- cgit v1.2.3 From 07573534b0b030226ee5ab560e53aac7e6c0dd84 Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Thu, 2 Jul 2009 11:57:19 -0600 Subject: ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleared When doing register reads, it is possible for there to be a stale data ready bit set which will cause subsequent reads to return prematurely with incorrect data. This patch fixes the issues by ensuring stale data is cleared before starting another transaction. Signed-off-by: Grant Likely Acked-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_psc_ac97.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 794a247b3eb..9b8503f2d68 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -41,6 +41,10 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) pr_err("timeout on ac97 bus (rdy)\n"); return -ENODEV; } + + /* Force clear the data valid bit */ + in_be32(&psc_dma->psc_regs->ac97_data); + /* Send the read */ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); -- cgit v1.2.3 From 0827d6ba0b76be398a3c4298afd41f4965d2cdcb Mon Sep 17 00:00:00 2001 From: Grant Likely Date: Thu, 2 Jul 2009 11:57:25 -0600 Subject: ASoC: add locking to mpc5200-psc-ac97 driver AC97 bus register read/write hooks need to provide locking, but the mpc5200-psc-ac97 driver does not. This patch adds a mutex around the register access routines. Signed-off-by: Grant Likely Acked-by: Jon Smirl Signed-off-by: Mark Brown --- sound/soc/fsl/mpc5200_dma.c | 1 + sound/soc/fsl/mpc5200_dma.h | 1 + sound/soc/fsl/mpc5200_psc_ac97.c | 13 ++++++++++++- 3 files changed, 14 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index efec33a1c5b..f0a2d407199 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -456,6 +456,7 @@ int mpc5200_audio_dma_create(struct of_device *op) return -ENODEV; spin_lock_init(&psc_dma->lock); + mutex_init(&psc_dma->mutex); psc_dma->id = be32_to_cpu(*prop); psc_dma->irq = irq; psc_dma->psc_regs = regs; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 2000803f06a..8d396bb9d9f 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -55,6 +55,7 @@ struct psc_dma { unsigned int irq; struct device *dev; spinlock_t lock; + struct mutex mutex; u32 sicr; uint sysclk; int imr; diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 9b8503f2d68..7eb549985d4 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -34,11 +34,14 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) int status; unsigned int val; + mutex_lock(&psc_dma->mutex); + /* Wait for command send status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } @@ -54,16 +57,19 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } /* Get the data */ val = in_be32(&psc_dma->psc_regs->ac97_data); if (((val >> 24) & 0x7f) != reg) { pr_err("reg echo error on ac97 read\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } val = (val >> 8) & 0xffff; + mutex_unlock(&psc_dma->mutex); return (unsigned short) val; } @@ -72,16 +78,21 @@ static void psc_ac97_write(struct snd_ac97 *ac97, { int status; + mutex_lock(&psc_dma->mutex); + /* Wait for command status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); - return; + goto out; } /* Write data */ out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + out: + mutex_unlock(&psc_dma->mutex); } static void psc_ac97_warm_reset(struct snd_ac97 *ac97) -- cgit v1.2.3 From 099db17e66294b02814dee01c81d9abbbeece93e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 2 Jul 2009 16:10:23 +0200 Subject: ALSA: hda - Add GPIO1 control at muting with HP laptops HP laptops with AD1984A codecs (at least mobile models) need to set GPIO1 appropriately to indicate the mute state. The BIOS checks this bit to judge whether the mute on or off is sent via F8 key. Without changing this bit, the BIOS can be confused and may toggle the mute wrongly. Reference: Novell bnc#515266 https://bugzilla.novell.com/show_bug.cgi?id=515266 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 85e8618e849..f795ee588cc 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -3734,9 +3734,30 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { } /* end */ }; +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(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + mute ? 0x02 : 0x0); + return ret; +} + static 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), + /*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .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("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), @@ -3857,6 +3878,10 @@ static struct hda_verb ad1884a_mobile_verbs[] = { /* 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 */ }; -- cgit v1.2.3 From aa202455eec51699e44f658530728162cefa1307 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 15:00:54 +0200 Subject: ALSA: hda - Improve ASUS eeePC 1000 mixer The mixer elements created for ASUS eeePC 1000 with ALC269 aren't standard but strange words like "LineOut". Rename the element names to follow the standard one like "Headphone" and "Speaker". Also, split the volumes to each so that the virtual master can control them. The alc269_fujitsu_mixer is removed because it's now identical with the new eeepc mixer. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 24 +++++------------------- 1 file changed, 5 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3a8e58c483d..e661b21354b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12876,20 +12876,11 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { } }; -/* bind volumes of both NID 0x0c and 0x0d */ -static struct hda_bind_ctls alc269_epc_bind_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT), - 0 - }, -}; - static struct snd_kcontrol_new alc269_eeepc_mixer[] = { - HDA_CODEC_MUTE("iSpeaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("LineOut Playback Volume", &alc269_epc_bind_vol), - HDA_CODEC_MUTE("LineOut Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -12902,12 +12893,7 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { }; /* FSC amilo */ -static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { - HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_BIND_VOL("PCM Playback Volume", &alc269_epc_bind_vol), - { } /* end */ -}; +#define alc269_fujitsu_mixer alc269_eeepc_mixer static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, -- cgit v1.2.3 From 637a935aaba2f05e2178c9d1b714d7a2c36c8b44 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 3 Jul 2009 01:04:16 +0200 Subject: ASoC: Fix wm8753 register cache size and initialization Register cache space was not being allocated for the final register, causing bugs when it was used. Allocate space for it. Also ensure that the final register is displayed in sysfs. [Commit message rewritten to document actual issue. -- broonie] Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index e06b0cfe4f2..49c4b2898af 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -79,7 +79,7 @@ static const u16 wm8753_reg[] = { 0x0097, 0x0097, 0x0000, 0x0004, 0x0000, 0x0083, 0x0024, 0x01ba, 0x0000, 0x0083, 0x0024, 0x01ba, - 0x0000, 0x0000 + 0x0000, 0x0000, 0x0000 }; /* codec private data */ @@ -1660,7 +1660,7 @@ static int wm8753_register(struct wm8753_priv *wm8753) codec->set_bias_level = wm8753_set_bias_level; codec->dai = wm8753_dai; codec->num_dai = 2; - codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache); + codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache) + 1; codec->reg_cache = &wm8753->reg_cache; codec->private_data = wm8753; -- cgit v1.2.3 From 022b466fc353d3dc7a152451144be656248666ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:03:30 +0200 Subject: ALSA: hda - Avoid invalid formats and rates with shared SPDIF Check whether formats and rates don't result in zero due to the restriction of SPDIF sharing. If any of them can be zero, disable the SPDIF sharing mode instead. Otherwise it will lead to a PCM configuration error. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 462e2cedaa6..26d255de6be 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3470,10 +3470,16 @@ int snd_hda_multi_out_analog_open(struct hda_codec *codec, } mutex_lock(&codec->spdif_mutex); if (mout->share_spdif) { - runtime->hw.rates &= mout->spdif_rates; - runtime->hw.formats &= mout->spdif_formats; - if (mout->spdif_maxbps < hinfo->maxbps) - hinfo->maxbps = mout->spdif_maxbps; + if ((runtime->hw.rates & mout->spdif_rates) && + (runtime->hw.formats & mout->spdif_formats)) { + runtime->hw.rates &= mout->spdif_rates; + runtime->hw.formats &= mout->spdif_formats; + if (mout->spdif_maxbps < hinfo->maxbps) + hinfo->maxbps = mout->spdif_maxbps; + } else { + mout->share_spdif = 0; + /* FIXME: need notify? */ + } } mutex_unlock(&codec->spdif_mutex); } -- cgit v1.2.3 From 70d321e6380f128096429d6e5b678f94ab0cef5d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:06:45 +0200 Subject: ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callback The PCM rates bit field may have been changed by the codec open callback. In that case, we need to reset rate_min and rate_max. So, simply call snd_pcm_lib_hw_rates() again after the codec open callback. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4e9ea708027..b36dc46615a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1454,6 +1454,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) mutex_unlock(&chip->open_mutex); return err; } + snd_pcm_limit_hw_rates(runtime); spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = substream; azx_dev->running = 0; -- cgit v1.2.3 From c470331e69bd54d11a9ea3c27a0e4ad783d02d6b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 3 Jul 2009 23:10:23 +0200 Subject: ALSA: hda - Add sanity check in PCM open callback Add some sanity checks of struct snd_pcm_hardware fields in the PCM open callback of hda driver. This makes a bit easier to debug any PCM setup errors in the codec side. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b36dc46615a..1877d95d4aa 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1464,6 +1464,12 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); + if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max)) + return -EINVAL; + if (snd_BUG_ON(!runtime->hw.formats)) + return -EINVAL; + if (snd_BUG_ON(!runtime->hw.rates)) + return -EINVAL; return 0; } -- cgit v1.2.3 From 954a973cab37ad5df3f87f08964166abd956cc17 Mon Sep 17 00:00:00 2001 From: Kay Sievers Date: Fri, 3 Jul 2009 20:56:05 +0200 Subject: sound: do not set DEVNAME for OSS devices Signed-off-by: Kay Sievers Signed-off-by: Takashi Iwai --- sound/sound_core.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/sound_core.c b/sound/sound_core.c index 12522e6913d..a41f8b127f4 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -10,6 +10,8 @@ #include #include #include +#include +#include #include #ifdef CONFIG_SOUND_OSS_CORE @@ -29,6 +31,8 @@ MODULE_LICENSE("GPL"); static char *sound_nodename(struct device *dev) { + if (MAJOR(dev->devt) == SOUND_MAJOR) + return NULL; return kasprintf(GFP_KERNEL, "snd/%s", dev_name(dev)); } @@ -104,7 +108,6 @@ module_exit(cleanup_soundcore); #include #include #include -#include #include #define SOUND_STEP 16 -- cgit v1.2.3 From 02358fcfa54ce018a0bb56ca9f5a898de574a9d3 Mon Sep 17 00:00:00 2001 From: Herton Ronaldo Krzesinski Date: Sat, 4 Jul 2009 01:44:59 -0300 Subject: ALSA: hda - move 8086:fb30 quirk (stac9205) to the proper section Signed-off-by: Herton Ronaldo Krzesinski Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 14f3c3e0f62..41b5b3a18c1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1590,8 +1590,6 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, - "SigmaTel",STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ @@ -2344,6 +2342,8 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xfb30, + "SigmaTel", STAC_9205_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_9205_REF), /* Dell */ -- cgit v1.2.3 From aba6653617754e12763a0d3c9dda332b66190a50 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 5 Jul 2009 11:44:46 +0200 Subject: ALSA: hda - Fix error path in the sanity check in azx_pcm_open() Release resources cleanly after errors in the sanity check in azx_pcm_open(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1877d95d4aa..16e09d74057 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1455,6 +1455,17 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) return err; } snd_pcm_limit_hw_rates(runtime); + /* sanity check */ + if (snd_BUG_ON(!runtime->hw.channels_min) || + snd_BUG_ON(!runtime->hw.channels_max) || + snd_BUG_ON(!runtime->hw.formats) || + snd_BUG_ON(!runtime->hw.rates)) { + azx_release_device(azx_dev); + hinfo->ops.close(hinfo, apcm->codec, substream); + snd_hda_power_down(apcm->codec); + mutex_unlock(&chip->open_mutex); + return -EINVAL; + } spin_lock_irqsave(&chip->reg_lock, flags); azx_dev->substream = substream; azx_dev->running = 0; @@ -1463,13 +1474,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->private_data = azx_dev; snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); - - if (snd_BUG_ON(!runtime->hw.channels_min || !runtime->hw.channels_max)) - return -EINVAL; - if (snd_BUG_ON(!runtime->hw.formats)) - return -EINVAL; - if (snd_BUG_ON(!runtime->hw.rates)) - return -EINVAL; return 0; } -- cgit v1.2.3 From 55d1d6c1ef630dddd3cb5354c32a5aca954399e8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Jul 2009 13:39:03 +0200 Subject: ALSA: hda - Clean up VT170x dig-in initialization code Minor clean up for initializing the digital-in pin. No functional changes. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 24 +++++++----------------- 1 file changed, 7 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 8e004fb6961..c4ddbbc6231 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -211,6 +211,7 @@ struct via_spec { unsigned int num_adc_nids; hda_nid_t *adc_nids; hda_nid_t dig_in_nid; + hda_nid_t dig_in_pin; /* capture source */ const struct hda_input_mux *input_mux; @@ -998,25 +999,11 @@ static int via_init(struct hda_codec *codec) /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ - if (IS_VT1708_VENDORID(codec->vendor_id)) { - snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, + if (spec->dig_in_pin) { + snd_hda_codec_write(codec, spec->dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, - AC_VERB_SET_EAPD_BTLENABLE, 0x02); - } else if (IS_VT1709_10CH_VENDORID(codec->vendor_id) || - IS_VT1709_6CH_VENDORID(codec->vendor_id)) { - snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_OUT); - snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, - AC_VERB_SET_EAPD_BTLENABLE, 0x02); - } else if (IS_VT1708B_8CH_VENDORID(codec->vendor_id) || - IS_VT1708B_4CH_VENDORID(codec->vendor_id)) { - snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - PIN_OUT); - snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, + snd_hda_codec_write(codec, spec->dig_in_pin, 0, AC_VERB_SET_EAPD_BTLENABLE, 0x02); } } else /* enable SPDIF-input pin */ @@ -1326,6 +1313,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708_DIGOUT_NID; + spec->dig_in_pin = VT1708_DIGIN_PIN; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708_DIGIN_NID; @@ -1799,6 +1787,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1709_DIGOUT_NID; + spec->dig_in_pin = VT1709_DIGIN_PIN; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1709_DIGIN_NID; @@ -2344,6 +2333,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID; + spec->dig_in_pin = VT1708B_DIGIN_PIN; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708B_DIGIN_NID; -- cgit v1.2.3 From d3a11e601a51291fbdd40c47f6af6769b6e905ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Jul 2009 13:43:35 +0200 Subject: ALSA: hda - Add missing EAPD initialization for VIA codecs If the output pin is used and EAPD capability is present, turn on the EAPD bit. This fixes the silent output problem on ASUS laptops with VT1708S codec. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c4ddbbc6231..322e1027247 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -320,6 +320,9 @@ static void via_auto_set_output_and_unmute(struct hda_codec *codec, pin_type); snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); + if (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_EAPD) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x02); } -- cgit v1.2.3 From 337b9d02b4873ceac91565272545fb6fd446d939 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Jul 2009 18:18:59 +0200 Subject: ALSA: hda - Fix capture source selection in patch_via.c The fixed widget NIDs in patch_via.c seem wrong for some codecs, and it resulted in the invalid capture source selection. This patch adds the code to parse the topology instead of using fixed numbers in order to get the right MUX widget id corresponding to the ADCs. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 54 ++++++++++++++++++++++++++++++----------------- 1 file changed, 35 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 322e1027247..38db4596422 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -210,6 +210,7 @@ struct via_spec { /* capture */ unsigned int num_adc_nids; hda_nid_t *adc_nids; + hda_nid_t mux_nids[3]; hda_nid_t dig_in_nid; hda_nid_t dig_in_pin; @@ -393,25 +394,11 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int vendor_id = codec->vendor_id; - /* AIW0 lydia 060801 add for correct sw0 input select */ - if (IS_VT1708_VENDORID(vendor_id) && (adc_idx == 0)) - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - 0x18, &spec->cur_mux[adc_idx]); - else if ((IS_VT1709_10CH_VENDORID(vendor_id) || - IS_VT1709_6CH_VENDORID(vendor_id)) && (adc_idx == 0)) - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - 0x19, &spec->cur_mux[adc_idx]); - else if ((IS_VT1708B_8CH_VENDORID(vendor_id) || - IS_VT1708B_4CH_VENDORID(vendor_id)) && (adc_idx == 0)) - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - 0x17, &spec->cur_mux[adc_idx]); - else if (IS_VT1702_VENDORID(vendor_id) && (adc_idx == 0)) - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - 0x13, &spec->cur_mux[adc_idx]); - else - return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, - spec->adc_nids[adc_idx], - &spec->cur_mux[adc_idx]); + if (!spec->mux_nids[adc_idx]) + return -EINVAL; + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + spec->mux_nids[adc_idx], + &spec->cur_mux[adc_idx]); } static int via_independent_hp_info(struct snd_kcontrol *kcontrol, @@ -1343,6 +1330,29 @@ static int via_auto_init(struct hda_codec *codec) return 0; } +static int get_mux_nids(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + hda_nid_t nid, conn[8]; + unsigned int type; + int i, n; + + for (i = 0; i < spec->num_adc_nids; i++) { + nid = spec->adc_nids[i]; + while (nid) { + n = snd_hda_get_connections(codec, nid, conn, + ARRAY_SIZE(conn)); + if (n <= 0) + break; + if (n > 1) { + spec->mux_nids[i] = nid; + break; + } + nid = conn[0]; + } + } +} + static int patch_vt1708(struct hda_codec *codec) { struct via_spec *spec; @@ -1851,6 +1861,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1709_capture_mixer; spec->num_mixers++; } @@ -1944,6 +1955,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1709_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1709_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1709_capture_mixer; spec->num_mixers++; } @@ -2397,6 +2409,7 @@ static int patch_vt1708B_8ch(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708B_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1708B_capture_mixer; spec->num_mixers++; } @@ -2448,6 +2461,7 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708B_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708B_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1708B_capture_mixer; spec->num_mixers++; } @@ -2882,6 +2896,7 @@ static int patch_vt1708S(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1708S_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1708S_capture_mixer; spec->num_mixers++; } @@ -3199,6 +3214,7 @@ static int patch_vt1702(struct hda_codec *codec) if (!spec->adc_nids && spec->input_mux) { spec->adc_nids = vt1702_adc_nids; spec->num_adc_nids = ARRAY_SIZE(vt1702_adc_nids); + get_mux_nids(codec); spec->mixers[spec->num_mixers] = vt1702_capture_mixer; spec->num_mixers++; } -- cgit v1.2.3 From 1c55d521f4e58be55735d7ac47e8197d6791fa9a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Jul 2009 07:45:46 +0200 Subject: ALSA: hda - Check widget types while parsing capture source in patch_via.c Check the widget type and don't take invalid widgets while parsing the capture source in patch_via.c. Also, fixed some compile warnings introduced in the previous commit. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 38db4596422..9008b4b013a 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -392,7 +392,6 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - unsigned int vendor_id = codec->vendor_id; if (!spec->mux_nids[adc_idx]) return -EINVAL; @@ -1340,6 +1339,10 @@ static int get_mux_nids(struct hda_codec *codec) for (i = 0; i < spec->num_adc_nids; i++) { nid = spec->adc_nids[i]; while (nid) { + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; + if (type == AC_WID_PIN) + break; n = snd_hda_get_connections(codec, nid, conn, ARRAY_SIZE(conn)); if (n <= 0) @@ -1351,6 +1354,7 @@ static int get_mux_nids(struct hda_codec *codec) nid = conn[0]; } } + return 0; } static int patch_vt1708(struct hda_codec *codec) -- cgit v1.2.3 From dc4c2e6bde77735071dbef7aca6bd6c0116102b3 Mon Sep 17 00:00:00 2001 From: Andiry Brienza Date: Wed, 8 Jul 2009 13:55:31 +0800 Subject: ALSA: hda - Disable AMD SB600 64bit address support only HDA driver disabled HD audio 64bit address support for all AMD SB600/SB700/SB800 platforms with commit 09240cf429505891d6123ce14a29f58f2a60121e due to one SB600 issue reported by community, but we do not see the similar issue on SB700/SB800 platforms. This patch is to refine the workaround for SB600 only. Signed-off-by: Andiry Xu Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 16e09d74057..77c1b840ca8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2333,9 +2333,19 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap = azx_readw(chip, GCAP); snd_printdd(SFX "chipset global capabilities = 0x%x\n", gcap); - /* ATI chips seems buggy about 64bit DMA addresses */ - if (chip->driver_type == AZX_DRIVER_ATI) - gcap &= ~ICH6_GCAP_64OK; + /* disable SB600 64bit support for safety */ + if ((chip->driver_type == AZX_DRIVER_ATI) || + (chip->driver_type == AZX_DRIVER_ATIHDMI)) { + struct pci_dev *p_smbus; + p_smbus = pci_get_device(PCI_VENDOR_ID_ATI, + PCI_DEVICE_ID_ATI_SBX00_SMBUS, + NULL); + if (p_smbus) { + if (p_smbus->revision < 0x30) + gcap &= ~ICH6_GCAP_64OK; + pci_dev_put(p_smbus); + } + } /* allow 64bit DMA address if supported by H/W */ if ((gcap & ICH6_GCAP_64OK) && !pci_set_dma_mask(pci, DMA_BIT_MASK(64))) -- cgit v1.2.3 From 508f711090e06477081fd94cb9298b1b14dda9ff Mon Sep 17 00:00:00 2001 From: Darren Salt Date: Wed, 8 Jul 2009 15:29:49 +0100 Subject: ALSA: hda - Missing volume controls for Intel HDA (ALC269/EeePC) There is a regression, introduced in aa202455eec51699e44f658530728162cefa1307 (in alsa-kernel) which I noticed when trying to use the headphone socket on my EeeCPC 901: the output was *very* quiet, practically silent. This patch corrects the control types to that which was obviously intended in the referenced commit. Signed-off-by: Darren Salt Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e661b21354b..c6c3d4a4d64 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12878,9 +12878,9 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { static struct snd_kcontrol_new alc269_eeepc_mixer[] = { HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), { } /* end */ }; -- cgit v1.2.3 From ad361c9884e809340f6daca80d56a9e9c871690a Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Mon, 6 Jul 2009 13:05:40 -0700 Subject: Remove multiple KERN_ prefixes from printk formats Commit 5fd29d6ccbc98884569d6f3105aeca70858b3e0f ("printk: clean up handling of log-levels and newlines") changed printk semantics. printk lines with multiple KERN_ prefixes are no longer emitted as before the patch. is now included in the output on each additional use. Remove all uses of multiple KERN_s in formats. Signed-off-by: Joe Perches Signed-off-by: Linus Torvalds --- sound/pci/emu10k1/p16v.c | 2 +- sound/usb/usx2y/usbusx2yaudio.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index e617acaf10e..61b8ab39800 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -644,7 +644,7 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * int err; int capture=1; - /* snd_printk("KERN_DEBUG snd_p16v_pcm called. device=%d\n", device); */ + /* snd_printk(KERN_DEBUG "snd_p16v_pcm called. device=%d\n", device); */ emu->p16v_device_offset = device; if (rpcm) *rpcm = NULL; diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index dd1ab617784..9efd27f6b52 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -296,9 +296,10 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, struct snd_usX2Y_substream *subs, struct urb *urb) { - snd_printk(KERN_ERR "Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" - KERN_ERR "Most propably some urb of usb-frame %i is still missing.\n" - KERN_ERR "Cause could be too long delays in usb-hcd interrupt handling.\n", + snd_printk(KERN_ERR +"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" +"Most propably some urb of usb-frame %i is still missing.\n" +"Cause could be too long delays in usb-hcd interrupt handling.\n", usb_get_current_frame_number(usX2Y->chip.dev), subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); -- cgit v1.2.3 From 369693dc93533097c0ca7243affb4f3244c336e8 Mon Sep 17 00:00:00 2001 From: Paul Vojta Date: Wed, 8 Jul 2009 23:57:46 -0700 Subject: ALSA: hda - fix beep tone calculation for IDT/STAC codecs In the beep tone calculation for IDT/STAC codecs, lower numbers correspond to higher frequencies and vice versa. The current code has this backwards, resulting in beep frequencies which are way too high (and sound bad on tinny laptop speakers, resulting in complaints). [Also added hz <= 0 check by tiwai] Signed-off-by: Paul Vojta Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 29272f2e95a..b0275a05087 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -50,19 +50,22 @@ static void snd_hda_generate_beep(struct work_struct *work) * The tone frequency of beep generator on IDT/STAC codecs is * defined from the 8bit tone parameter, in Hz, * freq = 48000 * (257 - tone) / 1024 - * that is from 12kHz to 93.75kHz in step of 46.875 hz + * that is from 12kHz to 93.75Hz in steps of 46.875 Hz */ static int beep_linear_tone(struct hda_beep *beep, int hz) { + if (hz <= 0) + return 0; hz *= 1000; /* fixed point */ - hz = hz - DIGBEEP_HZ_MIN; + hz = hz - DIGBEEP_HZ_MIN + + DIGBEEP_HZ_STEP / 2; /* round to nearest step */ if (hz < 0) hz = 0; /* turn off PC beep*/ else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) - hz = 0xff; + hz = 1; /* max frequency */ else { hz /= DIGBEEP_HZ_STEP; - hz++; + hz = 255 - hz; } return hz; } -- cgit v1.2.3 From d7dbf6ea40a2859adaca2dfdbbea83f3d6c73c2f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jul 2009 21:12:20 +0100 Subject: [ARM] 5596/1: at91sam9g20-ek: Register WM8731 in board file The WM8731 driver has been updated to allow registration via normal device model methods rather than from within the ASoC driver probe so update the AT91SAM9G20-EK to make use of this. Signed-off-by: Mark Brown Acked-by: Andrew Victor Signed-off-by: Russell King --- sound/soc/atmel/sam9g20_wm8731.c | 36 ------------------------------------ 1 file changed, 36 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 173a239a541..bb0db55e0e9 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -281,38 +281,6 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .set_bias_level = at91sam9g20ek_set_bias_level, }; -/* - * FIXME: This is a temporary bodge to avoid cross-tree merge issues. - * New drivers should register the wm8731 I2C device in the machine - * setup code (under arch/arm for ARM systems). - */ -static int wm8731_i2c_register(void) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = 0x1b; - strlcpy(info.type, "wm8731", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(0); - if (!adapter) { - printk(KERN_ERR "can't get i2c adapter 0\n"); - return -ENODEV; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - printk(KERN_ERR "can't add i2c device at 0x%x\n", - (unsigned int)info.addr); - return -ENODEV; - } - - return 0; -} - static struct snd_soc_device at91sam9g20ek_snd_devdata = { .card = &snd_soc_at91sam9g20ek, .codec_dev = &soc_codec_dev_wm8731, @@ -367,10 +335,6 @@ static int __init at91sam9g20ek_init(void) } ssc_p->ssc = ssc; - ret = wm8731_i2c_register(); - if (ret != 0) - goto err_ssc; - at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g20ek_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); -- cgit v1.2.3 From 005b10769c05fb16db70f7689ffb5ba17e3fc324 Mon Sep 17 00:00:00 2001 From: David Heidelberger Date: Thu, 9 Jul 2009 18:45:46 +0200 Subject: ALSA: hda - targa and targa-2ch fix Simplify ALC882_TARGA and return gpio3 to ALC883_TARGA_DIG and ALC883_TARGA_2ch_DIG, which I accidentally removed in commit id 64a8be74357477558183b43156c5536b642de134 Signed-off-by: David Heidelberger Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c6c3d4a4d64..bbb9b42e260 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6919,9 +6919,6 @@ static struct hda_verb alc882_targa_verbs[] = { {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, - {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x03}, - {0x01, AC_VERB_SET_GPIO_DATA, 0x03}, { } /* end */ }; @@ -7241,7 +7238,8 @@ static struct alc_config_preset alc882_presets[] = { }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer }, - .init_verbs = { alc882_init_verbs, alc882_targa_verbs}, + .init_verbs = { alc882_init_verbs, alc880_gpio3_init_verbs, + alc882_targa_verbs}, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .dig_out_nid = ALC882_DIGOUT_NID, @@ -9238,7 +9236,8 @@ static struct alc_config_preset alc883_presets[] = { }, [ALC883_TARGA_DIG] = { .mixers = { alc883_targa_mixer, alc883_chmode_mixer }, - .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, @@ -9251,7 +9250,8 @@ static struct alc_config_preset alc883_presets[] = { }, [ALC883_TARGA_2ch_DIG] = { .mixers = { alc883_targa_2ch_mixer}, - .init_verbs = { alc883_init_verbs, alc883_targa_verbs}, + .init_verbs = { alc883_init_verbs, alc880_gpio3_init_verbs, + alc883_targa_verbs}, .num_dacs = ARRAY_SIZE(alc883_dac_nids), .dac_nids = alc883_dac_nids, .adc_nids = alc883_adc_nids_alt, -- cgit v1.2.3