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(-) 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 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(-) 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 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(-) 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 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(-) 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 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(-) 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 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(-) 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 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(-) 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 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(-) 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 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(-) 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 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(-) 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 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(-) 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 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(-) 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