From eaeae5d9b783a62e435645122bed90561924a2d6 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 30 Sep 2009 09:27:24 +0300 Subject: ASoC: Fix SND_SOC_DAPM_LINE handling Since the SND_SOC_DAPM_LINE can be input or output, additional check is needed in order to determine if the widget is connected as input or output. When checking for connected outputs, if the widget is line, than check if the sources list is not empty (line is connected as output) For input endpoint check, when the widget is line, also check if the sinks list is not empty (line is connected as input). Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f79711b9fa5..8de6f9dec4a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -524,7 +524,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) /* connected jack or spk ? */ if (widget->id == snd_soc_dapm_hp || widget->id == snd_soc_dapm_spk || - widget->id == snd_soc_dapm_line) + (widget->id == snd_soc_dapm_line && !list_empty(&widget->sources))) return 1; } @@ -573,7 +573,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) return 1; /* connected jack ? */ - if (widget->id == snd_soc_dapm_mic || widget->id == snd_soc_dapm_line) + if (widget->id == snd_soc_dapm_mic || + (widget->id == snd_soc_dapm_line && !list_empty(&widget->sinks))) return 1; } -- cgit v1.2.3-70-g09d2 From e655a43544bd3c45a83da93b00a4b115b4fa758e Mon Sep 17 00:00:00 2001 From: Jonathan Cameron Date: Fri, 2 Oct 2009 16:09:49 +0100 Subject: ASoC: wm8940: Fix check on error code form snd_soc_codec_set_cache_io Fix for typo in commit 8d50e447d19fec64adebeef55f2b60d695435412 ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs Signed-off-by: Jonathan Cameron Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index da97aae475a..1ef2454c520 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -790,7 +790,7 @@ static int wm8940_register(struct wm8940_priv *wm8940, codec->reg_cache = &wm8940->reg_cache; ret = snd_soc_codec_set_cache_io(codec, 8, 16, control); - if (ret == 0) { + if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } -- cgit v1.2.3-70-g09d2 From 15870f05e90a365f8022da416e713be0c5024e2f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Oct 2009 08:25:13 +0200 Subject: ALSA: hda - Fix invalid initializations for ALC861 auto mode The recent auto-parser doesn't work for machines with a single output with ALC861, such as Toshiba laptops, because alc_subsystem_id() sets the hp_pins[0] while it's listed in line_outs[0]. This ends up with the doubled initialization of the same mixer widget, and it mutes the DAC route because hp_pins has no DAC assigned. To fix this problem, just check spec->autocfg.hp_outs and speaker_outs so that they are really detected pins. Reference: Novell bnc#544161 http://bugzilla.novell.com/show_bug.cgi?id=544161 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 15 ++++++++------- 1 file changed, 8 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7810d3dcad8..c1e05994cc3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14357,15 +14357,16 @@ static void alc861_auto_init_multi_out(struct hda_codec *codec) static void alc861_auto_init_hp_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - hda_nid_t pin; - pin = spec->autocfg.hp_pins[0]; - if (pin) - alc861_auto_set_output_and_unmute(codec, pin, PIN_HP, + if (spec->autocfg.hp_outs) + alc861_auto_set_output_and_unmute(codec, + spec->autocfg.hp_pins[0], + PIN_HP, spec->multiout.hp_nid); - pin = spec->autocfg.speaker_pins[0]; - if (pin) - alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, + if (spec->autocfg.speaker_outs) + alc861_auto_set_output_and_unmute(codec, + spec->autocfg.speaker_pins[0], + PIN_OUT, spec->multiout.dac_nids[0]); } -- cgit v1.2.3-70-g09d2 From f8f25ba3563dab14b1c3ea4d829642b8a61ca5d7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Oct 2009 08:31:29 +0200 Subject: ALSA: hda - Add a workaround for ASUS A7K ASUS A7K needs additional GPIO1 bit setup; it has to be cleared. Added a new fixup hook for this laptop so that it works as is. Refernece: Novell bnc#494309 http://bugzilla.novell.com/show_bug.cgi?id=494309 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 59 +++++++++++++++++++++++++++++++++++-------- 1 file changed, 48 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1e05994cc3..901c2999ed6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1362,7 +1362,7 @@ static void alc_ssid_check(struct hda_codec *codec, } /* - * Fix-up pin default configurations + * Fix-up pin default configurations and add default verbs */ struct alc_pincfg { @@ -1370,9 +1370,14 @@ struct alc_pincfg { u32 val; }; -static void alc_fix_pincfg(struct hda_codec *codec, +struct alc_fixup { + const struct alc_pincfg *pins; + const struct hda_verb *verbs; +}; + +static void alc_pick_fixup(struct hda_codec *codec, const struct snd_pci_quirk *quirk, - const struct alc_pincfg **pinfix) + const struct alc_fixup *fix) { const struct alc_pincfg *cfg; @@ -1380,9 +1385,14 @@ static void alc_fix_pincfg(struct hda_codec *codec, if (!quirk) return; - cfg = pinfix[quirk->value]; - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); + fix += quirk->value; + cfg = fix->pins; + if (cfg) { + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); + } + if (fix->verbs) + add_verb(codec->spec, fix->verbs); } /* @@ -9593,11 +9603,13 @@ static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { { } }; -static const struct alc_pincfg *alc882_pin_fixes[] = { - [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, +static const struct alc_fixup alc882_fixups[] = { + [PINFIX_ABIT_AW9D_MAX] = { + .pins = alc882_abit_aw9d_pinfix + }, }; -static struct snd_pci_quirk alc882_pinfix_tbl[] = { +static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), {} }; @@ -9869,7 +9881,7 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC882_AUTO; } - alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); + alc_pick_fixup(codec, alc882_fixup_tbl, alc882_fixups); if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ @@ -15159,7 +15171,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x103c, 0x30bf, "HP TX1000", ALC861VD_HP), SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), + /*SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST),*/ /* auto */ SND_PCI_QUIRK(0x1043, 0x1633, "Asus V1Sn", ALC660VD_ASUS_V1S), SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), @@ -15552,6 +15564,29 @@ static void alc861vd_auto_init(struct hda_codec *codec) alc_inithook(codec); } +enum { + ALC660VD_FIX_ASUS_GPIO1 +}; + +/* reset GPIO1 */ +static const struct hda_verb alc660vd_fix_asus_gpio1_verbs[] = { + {0x01, AC_VERB_SET_GPIO_MASK, 0x03}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + { } +}; + +static const struct alc_fixup alc861vd_fixups[] = { + [ALC660VD_FIX_ASUS_GPIO1] = { + .verbs = alc660vd_fix_asus_gpio1_verbs, + }, +}; + +static struct snd_pci_quirk alc861vd_fixup_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1339, "ASUS A7-K", ALC660VD_FIX_ASUS_GPIO1), + {} +}; + static int patch_alc861vd(struct hda_codec *codec) { struct alc_spec *spec; @@ -15573,6 +15608,8 @@ static int patch_alc861vd(struct hda_codec *codec) board_config = ALC861VD_AUTO; } + alc_pick_fixup(codec, alc861vd_fixup_tbl, alc861vd_fixups); + if (board_config == ALC861VD_AUTO) { /* automatic parse from the BIOS config */ err = alc861vd_parse_auto_config(codec); -- cgit v1.2.3-70-g09d2 From 01d4825df62d1d405035b90294bf38616d3f380b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 6 Oct 2009 13:21:54 +0200 Subject: ALSA: hda - Don't pick up invalid HP pins in alc_subsystem_id() alc_subsystem_id() tries to pick up a headphone pin if not configured, but this caused side-effects as the problem in commit 15870f05e90a365f8022da416e713be0c5024e2f. This patch fixes the driver behavior to pick up invalid HP pins; at least, the pins that are listed as the primary outputs aren't taken any more. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 901c2999ed6..a61fbbb41b2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1332,15 +1332,20 @@ do_sku: * when the external headphone out jack is plugged" */ if (!spec->autocfg.hp_pins[0]) { + hda_nid_t nid; tmp = (ass >> 11) & 0x3; /* HP to chassis */ if (tmp == 0) - spec->autocfg.hp_pins[0] = porta; + nid = porta; else if (tmp == 1) - spec->autocfg.hp_pins[0] = porte; + nid = porte; else if (tmp == 2) - spec->autocfg.hp_pins[0] = portd; + nid = portd; else return 1; + for (i = 0; i < spec->autocfg.line_outs; i++) + if (spec->autocfg.line_out_pins[i] == nid) + return 1; + spec->autocfg.hp_pins[0] = nid; } alc_init_auto_hp(codec); -- cgit v1.2.3-70-g09d2 From 2fb930b53f513cbc4c102d415d2923a8a7091337 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 6 Oct 2009 08:21:04 +0200 Subject: sound: via82xx: move DXS volume controls to PCM interface The "VIA DXS" controls are actually volume controls that apply to the four PCM substreams, so we better indicate this connection by moving the controls to the PCM interface. Commit b452e08e73c0e3dbb0be82130217be4b7084299e in 2.6.30 broke the restoring of these volumes by "alsactl restore" that most distributions use; the renaming in this patch cures that regression by preventing alsactl from applying the old, wrong volume levels to the new controls. http://bugzilla.kernel.org/show_bug.cgi?id=14151 http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=532613 Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/pci/via82xx.c | 27 ++++++++++++++++++--------- 1 file changed, 18 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index acfa4760da4..91683a34903 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -1626,7 +1626,7 @@ static int snd_via8233_dxs_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct via82xx *chip = snd_kcontrol_chip(kcontrol); - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); + unsigned int idx = kcontrol->id.subdevice; ucontrol->value.integer.value[0] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][0]; ucontrol->value.integer.value[1] = VIA_DXS_MAX_VOLUME - chip->playback_volume[idx][1]; @@ -1646,7 +1646,7 @@ static int snd_via8233_dxs_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct via82xx *chip = snd_kcontrol_chip(kcontrol); - unsigned int idx = snd_ctl_get_ioff(kcontrol, &ucontrol->id); + unsigned int idx = kcontrol->id.subdevice; unsigned long port = chip->port + 0x10 * idx; unsigned char val; int i, change = 0; @@ -1705,11 +1705,12 @@ static struct snd_kcontrol_new snd_via8233_pcmdxs_volume_control __devinitdata = }; static struct snd_kcontrol_new snd_via8233_dxs_volume_control __devinitdata = { - .name = "VIA DXS Playback Volume", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .device = 0, + /* .subdevice set later */ + .name = "PCM Playback Volume", .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ), - .count = 4, .info = snd_via8233_dxs_volume_info, .get = snd_via8233_dxs_volume_get, .put = snd_via8233_dxs_volume_put, @@ -1936,10 +1937,18 @@ static int __devinit snd_via8233_init_misc(struct via82xx *chip) } else /* Using DXS when PCM emulation is enabled is really weird */ { - /* Standalone DXS controls */ - err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_via8233_dxs_volume_control, chip)); - if (err < 0) - return err; + for (i = 0; i < 4; ++i) { + struct snd_kcontrol *kctl; + + kctl = snd_ctl_new1( + &snd_via8233_dxs_volume_control, chip); + if (!kctl) + return -ENOMEM; + kctl->id.subdevice = i; + err = snd_ctl_add(chip->card, kctl); + if (err < 0) + return err; + } } } /* select spdif data slot 10/11 */ -- cgit v1.2.3-70-g09d2 From b266002abf6dfa4b358fdb5495f09e350b296552 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 6 Oct 2009 19:25:02 +0100 Subject: ASoC: Remove absent SYNC and TDM DAI format options from i.MX SSI These should be handled via set_tdm_slot() now and cause build failures as-is. Signed-off-by: Mark Brown --- sound/soc/imx/mxc-ssi.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/mxc-ssi.c b/sound/soc/imx/mxc-ssi.c index 3806ff2c0cd..ccdefe60e75 100644 --- a/sound/soc/imx/mxc-ssi.c +++ b/sound/soc/imx/mxc-ssi.c @@ -397,14 +397,6 @@ static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; } - /* sync */ - if (!(fmt & SND_SOC_DAIFMT_ASYNC)) - scr |= SSI_SCR_SYN; - - /* tdm - only for stereo atm */ - if (fmt & SND_SOC_DAIFMT_TDM) - scr |= SSI_SCR_NET; - if (cpu_dai->id == IMX_DAI_SSI0 || cpu_dai->id == IMX_DAI_SSI2) { SSI1_STCR = stcr; SSI1_SRCR = srcr; -- cgit v1.2.3-70-g09d2 From 5b7dde346881b12246669ae97b3a2793c27b32b6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Jun 2009 11:17:10 +0100 Subject: ASoC: WM8350 capture PGA mutes are inverted Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/wm8350.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e7348d341b7..26f826c6e74 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -580,7 +580,7 @@ static const struct snd_kcontrol_new wm8350_left_capt_mixer_controls[] = { SOC_DAPM_SINGLE_TLV("L3 Capture Volume", WM8350_INPUT_MIXER_VOLUME_L, 9, 7, 0, out_mix_tlv), SOC_DAPM_SINGLE("PGA Capture Switch", - WM8350_LEFT_INPUT_VOLUME, 14, 1, 0), + WM8350_LEFT_INPUT_VOLUME, 14, 1, 1), }; /* Right Input Mixer */ @@ -590,7 +590,7 @@ static const struct snd_kcontrol_new wm8350_right_capt_mixer_controls[] = { SOC_DAPM_SINGLE_TLV("L3 Capture Volume", WM8350_INPUT_MIXER_VOLUME_R, 13, 7, 0, out_mix_tlv), SOC_DAPM_SINGLE("PGA Capture Switch", - WM8350_RIGHT_INPUT_VOLUME, 14, 1, 0), + WM8350_RIGHT_INPUT_VOLUME, 14, 1, 1), }; /* Left Mic Mixer */ -- cgit v1.2.3-70-g09d2 From defb5ab2e0ff08ff9a942e2bb7e14c21a55ec26b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 7 Oct 2009 15:12:27 +0200 Subject: ALSA: hda - Fix yet another auto-mic bug in ALC268 Since patch_alc268() doesn't call set_capture_mixer() (due to its h/w design different from other siblings), it needs to call fixup_automic_adc() explicitly to set up the auto-mic routing. Otherwise the indices for int/ext mics aren't set properly. Reference: Novell bnc#544899 http://bugzilla.novell.com/show_bug.cgi?id=544899 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a61fbbb41b2..470fd74a0a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12859,12 +12859,15 @@ static int patch_alc268(struct hda_codec *codec) unsigned int wcap = get_wcaps(codec, 0x07); int i; + spec->capsrc_nids = alc268_capsrc_nids; /* get type */ wcap = get_wcaps_type(wcap); if (spec->auto_mic || wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) { spec->adc_nids = alc268_adc_nids_alt; spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt); + if (spec->auto_mic) + fixup_automic_adc(codec); if (spec->auto_mic || spec->input_mux->num_items == 1) add_mixer(spec, alc268_capture_nosrc_mixer); else @@ -12874,7 +12877,6 @@ static int patch_alc268(struct hda_codec *codec) spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids); add_mixer(spec, alc268_capture_mixer); } - spec->capsrc_nids = alc268_capsrc_nids; /* set default input source */ for (i = 0; i < spec->num_adc_nids; i++) snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], -- cgit v1.2.3-70-g09d2 From 2bdf66331c3ff8d564efe7a054f1099133d520cd Mon Sep 17 00:00:00 2001 From: Pavel Hofman Date: Tue, 6 Oct 2009 16:04:11 +0200 Subject: ALSA: ICE1712/24 - Change the Multi Track Peak control (level meters) from MIXER to PCM type * PLEASE NOTE - this change requires the corresponding update of envy24control for ice1712 - kind of an ABI change. * The "Multi Track Peak" control is read-only level meters indicator. * The control is VERY confusing to most users since it is currently displayed in regular mixers. E.g. alsamixer ignores its read-only status and allows changing the levels with keys which makes no sense. Signed-off-by: Pavel Hofman Acked-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index cecf1ffeeaa..d74033a2cfb 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2259,7 +2259,7 @@ static int snd_ice1712_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_ice1712_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_ice1712_pro_peak_info, diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index af6e0014862..c24f268f63a 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -2110,7 +2110,7 @@ static int snd_vt1724_pro_peak_get(struct snd_kcontrol *kcontrol, } static struct snd_kcontrol_new snd_vt1724_mixer_pro_peak __devinitdata = { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, .name = "Multi Track Peak", .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, .info = snd_vt1724_pro_peak_info, -- cgit v1.2.3-70-g09d2 From 8dce39b8955be6164172cb6204ef8fc21de27431 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 7 Oct 2009 22:51:34 +0200 Subject: ALSA: opl3: circular locking in the snd_opl3_note_on() and snd_opl3_note_off() Fix following circular locking in the opl3 driver. ======================================================= [ INFO: possible circular locking dependency detected ] 2.6.32-rc3 #87 ------------------------------------------------------- swapper/0 is trying to acquire lock: (&opl3->voice_lock){..-...}, at: [] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] but task is already holding lock: (&opl3->sys_timer_lock){..-...}, at: [] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth] which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (&opl3->sys_timer_lock){..-...}: [] validate_chain+0xa25/0x1040 [] __lock_acquire+0x2da/0xab0 [] lock_acquire+0x7a/0xa0 [] _spin_lock_irqsave+0x40/0x60 [] snd_opl3_note_on+0x686/0x790 [snd_opl3_synth] [] snd_midi_process_event+0x322/0x590 [snd_seq_midi_emul] [] snd_opl3_synth_event_input+0x15/0x20 [snd_opl3_synth] [] snd_seq_deliver_single_event+0x100/0x200 [snd_seq] [] snd_seq_deliver_event+0x47/0x1f0 [snd_seq] [] snd_seq_dispatch_event+0x3b/0x140 [snd_seq] [] snd_seq_check_queue+0x10c/0x120 [snd_seq] [] snd_seq_enqueue_event+0x6b/0xe0 [snd_seq] [] snd_seq_client_enqueue_event+0xdd/0x100 [snd_seq] [] snd_seq_write+0xea/0x190 [snd_seq] [] vfs_write+0x96/0x160 [] sys_write+0x3d/0x70 [] syscall_call+0x7/0xb -> #0 (&opl3->voice_lock){..-...}: [] validate_chain+0x1036/0x1040 [] __lock_acquire+0x2da/0xab0 [] lock_acquire+0x7a/0xa0 [] _spin_lock_irqsave+0x40/0x60 [] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth] [] run_timer_softirq+0x166/0x1e0 [] __do_softirq+0x78/0x110 [] do_softirq+0x46/0x50 [] irq_exit+0x36/0x40 [] do_IRQ+0x42/0xb0 [] common_interrupt+0x2e/0x40 [] apm_cpu_idle+0x10f/0x290 [] cpu_idle+0x21/0x40 [] rest_init+0x4d/0x60 [] start_kernel+0x235/0x280 [] i386_start_kernel+0x66/0x70 other info that might help us debug this: 2 locks held by swapper/0: #0: (&opl3->tlist){+.-...}, at: [] run_timer_softirq+0xf0/0x1e0 #1: (&opl3->sys_timer_lock){..-...}, at: [] snd_opl3_timer_func+0x19/0xc0 [snd_opl3_synth] stack backtrace: Pid: 0, comm: swapper Not tainted 2.6.32-rc3 #87 Call Trace: [] print_circular_bug+0xc8/0xd0 [] validate_chain+0x1036/0x1040 [] ? check_usage_forwards+0x54/0xd0 [] __lock_acquire+0x2da/0xab0 [] lock_acquire+0x7a/0xa0 [] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [] _spin_lock_irqsave+0x40/0x60 [] ? snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [] snd_opl3_note_off+0x1e/0xe0 [snd_opl3_synth] [] ? _spin_lock_irqsave+0x47/0x60 [] snd_opl3_timer_func+0xa0/0xc0 [snd_opl3_synth] [] run_timer_softirq+0x166/0x1e0 [] ? run_timer_softirq+0xf0/0x1e0 [] ? snd_opl3_timer_func+0x0/0xc0 [snd_opl3_synth] [] __do_softirq+0x78/0x110 [] ? _spin_unlock+0x1d/0x20 [] ? handle_level_irq+0xaf/0xe0 [] do_softirq+0x46/0x50 [] irq_exit+0x36/0x40 [] do_IRQ+0x42/0xb0 [] ? trace_hardirqs_on_caller+0x12c/0x180 [] common_interrupt+0x2e/0x40 [] ? default_idle+0x38/0x50 [] apm_cpu_idle+0x10f/0x290 [] cpu_idle+0x21/0x40 [] rest_init+0x4d/0x60 [] start_kernel+0x235/0x280 [] ? unknown_bootoption+0x0/0x210 [] i386_start_kernel+0x66/0x70 Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/drivers/opl3/opl3_midi.c | 28 ++++++++++++++++++++-------- 1 file changed, 20 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 6e7d09ae0e8..7d722a025d0 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -29,6 +29,8 @@ extern char snd_opl3_regmap[MAX_OPL2_VOICES][4]; extern int use_internal_drums; +static void snd_opl3_note_off_unsafe(void *p, int note, int vel, + struct snd_midi_channel *chan); /* * The next table looks magical, but it certainly is not. Its values have * been calculated as table[i]=8*log(i/64)/log(2) with an obvious exception @@ -242,16 +244,20 @@ void snd_opl3_timer_func(unsigned long data) int again = 0; int i; - spin_lock_irqsave(&opl3->sys_timer_lock, flags); + spin_lock_irqsave(&opl3->voice_lock, flags); for (i = 0; i < opl3->max_voices; i++) { struct snd_opl3_voice *vp = &opl3->voices[i]; if (vp->state > 0 && vp->note_off_check) { if (vp->note_off == jiffies) - snd_opl3_note_off(opl3, vp->note, 0, vp->chan); + snd_opl3_note_off_unsafe(opl3, vp->note, 0, + vp->chan); else again++; } } + spin_unlock_irqrestore(&opl3->voice_lock, flags); + + spin_lock_irqsave(&opl3->sys_timer_lock, flags); if (again) { opl3->tlist.expires = jiffies + 1; /* invoke again */ add_timer(&opl3->tlist); @@ -658,15 +664,14 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice) /* * Release a note in response to a midi note off. */ -void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan) +static void snd_opl3_note_off_unsafe(void *p, int note, int vel, + struct snd_midi_channel *chan) { struct snd_opl3 *opl3; int voice; struct snd_opl3_voice *vp; - unsigned long flags; - opl3 = p; #ifdef DEBUG_MIDI @@ -674,12 +679,9 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan chan->number, chan->midi_program, note); #endif - spin_lock_irqsave(&opl3->voice_lock, flags); - if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) { if (chan->drum_channel && use_internal_drums) { snd_opl3_drum_switch(opl3, note, vel, 0, chan); - spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } /* this loop will hopefully kill all extra voices, because @@ -697,6 +699,16 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan snd_opl3_kill_voice(opl3, voice); } } +} + +void snd_opl3_note_off(void *p, int note, int vel, + struct snd_midi_channel *chan) +{ + struct snd_opl3 *opl3 = p; + unsigned long flags; + + spin_lock_irqsave(&opl3->voice_lock, flags); + snd_opl3_note_off_unsafe(p, note, vel, chan); spin_unlock_irqrestore(&opl3->voice_lock, flags); } -- cgit v1.2.3-70-g09d2 From 1d4efa6650454177afe30ad97283ff78572d0442 Mon Sep 17 00:00:00 2001 From: Robert Hancock Date: Wed, 7 Oct 2009 20:19:21 -0600 Subject: ALSA: ice1724: increase SPDIF and independent stereo buffer sizes Increase the default and maximum PCM buffer prellocation size for ice1724's SPDIF and independent stereo pair outputs to 256K, which is the hardware's maximum supported size. This allows a reduction in interrupt rate and potentially power usage when an application is not latency-critical. Signed-off-by: Robert Hancock Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index c24f268f63a..76b717dae4b 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -1294,7 +1294,7 @@ static int __devinit snd_vt1724_pcm_spdif(struct snd_ice1712 *ice, int device) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ice->pci), - 64*1024, 64*1024); + 256*1024, 256*1024); ice->pcm = pcm; @@ -1408,7 +1408,7 @@ static int __devinit snd_vt1724_pcm_indep(struct snd_ice1712 *ice, int device) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(ice->pci), - 64*1024, 64*1024); + 256*1024, 256*1024); ice->pcm_ds = pcm; -- cgit v1.2.3-70-g09d2 From f0613d5752d8f7d1d02e6d40947f38877fdf9c90 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Oct 2009 17:44:08 +0200 Subject: ALSA: hda - Add full rates/formats support for Nvidia HDMI Allow Nvidia HDMI to support more possible sample rates and formats. At best, the really supported rates and formats should be determined together with the negotiation with the HDMI receiver, but it's currently not implemented yet (Nvidia stuff seems incompatible with HDMI 1.3 standard in this regard). As a compromise, we enable all bits, assuming that all recent devices do support such rates/formats. Tested-by: Alan Alan Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 31 +++++++++++++++++++++++++------ 1 file changed, 25 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index c8435c9a97f..23ad9398311 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -29,6 +29,9 @@ #include "hda_codec.h" #include "hda_local.h" +/* define below to restrict the supported rates and formats */ +#define LIMITED_RATE_FMT_SUPPORT + struct nvhdmi_spec { struct hda_multi_out multiout; @@ -60,6 +63,22 @@ static struct hda_verb nvhdmi_basic_init[] = { {} /* terminator */ }; +#ifdef LIMITED_RATE_FMT_SUPPORT +/* support only the safe format and rate */ +#define SUPPORTED_RATES SNDRV_PCM_RATE_48000 +#define SUPPORTED_MAXBPS 16 +#define SUPPORTED_FORMATS SNDRV_PCM_FMTBIT_S16_LE +#else +/* support all rates and formats */ +#define SUPPORTED_RATES \ + (SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define SUPPORTED_MAXBPS 24 +#define SUPPORTED_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#endif + /* * Controls */ @@ -258,9 +277,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_8ch = { .channels_min = 2, .channels_max = 8, .nid = Nv_Master_Convert_nid, - .rates = SNDRV_PCM_RATE_48000, - .maxbps = 16, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SUPPORTED_RATES, + .maxbps = SUPPORTED_MAXBPS, + .formats = SUPPORTED_FORMATS, .ops = { .open = nvhdmi_dig_playback_pcm_open, .close = nvhdmi_dig_playback_pcm_close_8ch, @@ -273,9 +292,9 @@ static struct hda_pcm_stream nvhdmi_pcm_digital_playback_2ch = { .channels_min = 2, .channels_max = 2, .nid = Nv_Master_Convert_nid, - .rates = SNDRV_PCM_RATE_48000, - .maxbps = 16, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SUPPORTED_RATES, + .maxbps = SUPPORTED_MAXBPS, + .formats = SUPPORTED_FORMATS, .ops = { .open = nvhdmi_dig_playback_pcm_open, .close = nvhdmi_dig_playback_pcm_close_2ch, -- cgit v1.2.3-70-g09d2 From 43189a38dada053b820fafc47de8ba665dd3a618 Mon Sep 17 00:00:00 2001 From: Robert Hancock Date: Fri, 9 Oct 2009 22:08:58 -0600 Subject: ALSA: ice1724: Fix surround on Chaintech AV-710 Fix the num_total_dacs setting for Chaintech AV710. The existing comment that only PSDOUT0 is connected is correct, but since the card is using packed AC97 mode to send 6 channels to the codec, num_total_dacs should be set to 6 and not 2. This allows 6-channel surround to work. Also clarify a comment regarding the additional WM8728 codec on this card (it's connected to the SPDIF output and always receives the same data). Signed-off-by: Robert Hancock Signed-off-by: Takashi Iwai --- sound/pci/ice1712/amp.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index 37564300b50..6da21a2bcad 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -52,11 +52,13 @@ static int __devinit snd_vt1724_amp_init(struct snd_ice1712 *ice) /* only use basic functionality for now */ - ice->num_total_dacs = 2; /* only PSDOUT0 is connected */ + /* VT1616 6ch codec connected to PSDOUT0 using packed mode */ + ice->num_total_dacs = 6; ice->num_total_adcs = 2; - /* Chaintech AV-710 has another codecs, which need initialization */ - /* initialize WM8728 codec */ + /* Chaintech AV-710 has another WM8728 codec connected to PSDOUT4 + (shared with the SPDIF output). Mixer control for this codec + is not yet supported. */ if (ice->eeprom.subvendor == VT1724_SUBDEVICE_AV710) { for (i = 0; i < ARRAY_SIZE(wm_inits); i += 2) wm_put(ice, wm_inits[i], wm_inits[i+1]); -- cgit v1.2.3-70-g09d2 From bd3c200e6d5495343c91db66d2acf1853b57a141 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Sun, 11 Oct 2009 11:37:22 +0200 Subject: ALSA: ice1724 - Make call to set hw params succeed on ESI Juli@ If two streams are started immediately after one another (such as a playback and a recording stream), the call to set hw params fails with EBUSY. This patch makes the call succeed, so playback and recording will work properly. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1724.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 76b717dae4b..10fc92c0557 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -648,7 +648,7 @@ static int snd_vt1724_set_pro_rate(struct snd_ice1712 *ice, unsigned int rate, (inb(ICEMT1724(ice, DMA_PAUSE)) & DMA_PAUSES)) { /* running? we cannot change the rate now... */ spin_unlock_irqrestore(&ice->reg_lock, flags); - return -EBUSY; + return ((rate == ice->cur_rate) && !force) ? 0 : -EBUSY; } if (!force && is_pro_rate_locked(ice)) { spin_unlock_irqrestore(&ice->reg_lock, flags); -- cgit v1.2.3-70-g09d2 From 2d9c648295d7bc376305337d29f540a5e411f632 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Oct 2009 08:06:55 +0200 Subject: ALSA: hda - Fix overflow of spec->init_verbs in patch_realtek.c ALC861-VD lenovo model causes overflow of spec->init_verbs entries due to the recent changes. Simply increase the array size to avoid the overflow. Reported-by: Luca Tettamanti 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 470fd74a0a1..c08ca660dab 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -275,7 +275,7 @@ struct alc_spec { struct snd_kcontrol_new *cap_mixer; /* capture mixer */ unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ - const struct hda_verb *init_verbs[5]; /* initialization verbs + const struct hda_verb *init_verbs[10]; /* initialization verbs * don't forget NULL * termination! */ -- cgit v1.2.3-70-g09d2 From 9c6b8dcefe9a39f36ba11bdd523c0ac5246514c9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Oct 2009 09:34:28 +0200 Subject: ALSA: bt87x - Add a whitelist for Pinnacle PCTV (11bd:0012) Signed-off-by: Takashi Iwai --- sound/pci/bt87x.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 24585c6c6d0..4e2b925a94c 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -808,6 +808,8 @@ static struct pci_device_id snd_bt87x_ids[] = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, GENERIC), /* Leadtek Winfast tv 2000xp delux */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, GENERIC), + /* Pinnacle PCTV */ + BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x11bd, 0x0012, GENERIC), /* Voodoo TV 200 */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, GENERIC), /* Askey Computer Corp. MagicTView'99 */ -- cgit v1.2.3-70-g09d2 From 54930531a00af5a1c33361a02e67dd1802110465 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 11 Oct 2009 17:38:29 +0200 Subject: ALSA: hda - Fix mute sound with STAC9227/9228 codecs On FSC laptops, the sound gets muted gradually when the volume is chnaged. This is due to the wrong volume-knob widget setup. The delta bit (bit 7) shouldn't be set for these devices. This patch adds a new quirk to set the value 0x7f to the widget 0x24 instead of 0xff. Reference: Novell bnc#546006 http://bugzilla.novell.com/show_bug.cgi?id=546006 Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_sigmatel.c | 17 +++++++++++++++++ 2 files changed, 18 insertions(+) (limited to 'sound') diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index a2643cfe793..4bf953b55b8 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -359,6 +359,7 @@ STAC9227/9228/9229/927x 5stack-no-fp D965 5stack without front panel dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup + volknob Fixes with volume-knob widget 0x24 auto BIOS setup (default) STAC92HD71B* diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a9b26828a65..75736827425 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -158,6 +158,7 @@ enum { STAC_D965_5ST_NO_FP, STAC_DELL_3ST, STAC_DELL_BIOS, + STAC_927X_VOLKNOB, STAC_927X_MODELS }; @@ -915,6 +916,14 @@ static struct hda_verb stac927x_core_init[] = { {} }; +static struct hda_verb stac927x_volknob_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* enable analog pc beep path */ + {0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, + {} +}; + static struct hda_verb stac9205_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -1999,6 +2008,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { [STAC_D965_5ST_NO_FP] = d965_5st_no_fp_pin_configs, [STAC_DELL_3ST] = dell_3st_pin_configs, [STAC_DELL_BIOS] = NULL, + [STAC_927X_VOLKNOB] = NULL, }; static const char *stac927x_models[STAC_927X_MODELS] = { @@ -2010,6 +2020,7 @@ static const char *stac927x_models[STAC_927X_MODELS] = { [STAC_D965_5ST_NO_FP] = "5stack-no-fp", [STAC_DELL_3ST] = "dell-3stack", [STAC_DELL_BIOS] = "dell-bios", + [STAC_927X_VOLKNOB] = "volknob", }; static struct snd_pci_quirk stac927x_cfg_tbl[] = { @@ -2045,6 +2056,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500, "Intel D965", STAC_D965_5ST), + /* volume-knob fixes */ + SND_PCI_QUIRK_VENDOR(0x10cf, "FSC", STAC_927X_VOLKNOB), {} /* terminator */ }; @@ -5616,6 +5629,10 @@ static int patch_stac927x(struct hda_codec *codec) spec->dmux_nids = stac927x_dmux_nids; spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); break; + case STAC_927X_VOLKNOB: + spec->num_dmics = 0; + spec->init = stac927x_volknob_core_init; + break; default: spec->num_dmics = 0; spec->init = stac927x_core_init; -- cgit v1.2.3-70-g09d2 From ccca7cdc1b8dd2e7b67e9289a6abf117b11cbe6b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Oct 2009 15:32:21 +0200 Subject: ALSA: hda - Fix volume-knob setup for Dell laptops with STAC9228 The volume-knob widget needs to be set with 0x7f instead of 0xff for Dell laptops with STAC9228 codec, too, like the previous commit. Reference: Novell bnc#545013 http://bugzilla.novell.com/show_bug.cgi?id=545013 Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 75736827425..66c0876bf73 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -908,6 +908,16 @@ static struct hda_verb d965_core_init[] = { {} }; +static struct hda_verb dell_3st_core_init[] = { + /* don't set delta bit */ + {0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0x7f}, + /* unmute node 0x1b */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + /* select node 0x03 as DAC */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0x01}, + {} +}; + static struct hda_verb stac927x_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, @@ -5625,7 +5635,7 @@ static int patch_stac927x(struct hda_codec *codec) spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->init = d965_core_init; + spec->init = dell_3st_core_init; spec->dmux_nids = stac927x_dmux_nids; spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); break; -- cgit v1.2.3-70-g09d2 From 29a4f2d31c03756bf24883e567a8c3b4ee5df1f4 Mon Sep 17 00:00:00 2001 From: Philby John Date: Tue, 13 Oct 2009 16:30:22 +0530 Subject: ALSA: aaci: ARM1176 aaci-pl041 AC97 register read timeout After a reboot on an ARM1176 which amounts to a softreset, it has been noted that the ALSA driver does not get registered and the probe fails with the error "aaci-pl041 fpga:04: ac97 read back fail". In the process of reading from a register the SL1TxBusy bit is set indicating that the transceiver is busy and remains so until the default timeout occurs. Set the Power down register 0x26 to an arbitrary value as specified in the PL041 manual (page: 3-18) so that AACISL1TX/AACISL2TX registers take their default state. Signed-off-by: Philby John Signed-off-by: Takashi Iwai --- sound/arm/aaci.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index dc78272fc39..1f0f8213e2d 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -937,6 +937,7 @@ static int __devinit aaci_probe_ac97(struct aaci *aaci) struct snd_ac97 *ac97; int ret; + writel(0, aaci->base + AC97_POWERDOWN); /* * Assert AACIRESET for 2us */ -- cgit v1.2.3-70-g09d2 From 491dc0437d4c56d11f78113eca3953cff87314f3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 13 Oct 2009 16:07:59 +0200 Subject: ALSA: hda - Allow all formats as default for Nvidia HDMI In the commit f0613d5752d8f7d1d02e6d40947f38877fdf9c90 ALSA: hda - Add full rates/formats support for Nvidia HDMI the flag LIMITIED_RATE_FMT_SUPPORT was set as default, as I forgot to clear before commit. Let's enable all formats/rates as default. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 23ad9398311..9fb60276f5c 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -30,7 +30,7 @@ #include "hda_local.h" /* define below to restrict the supported rates and formats */ -#define LIMITED_RATE_FMT_SUPPORT +/* #define LIMITED_RATE_FMT_SUPPORT */ struct nvhdmi_spec { struct hda_multi_out multiout; -- cgit v1.2.3-70-g09d2