From 048e78a5bc22c27410cb5ca9680c3c7ac400607f Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 2 Sep 2010 08:35:47 +0200 Subject: ALSA: hda - Add a new hp-laptop model for Conexant 5066, tested on HP G60 This new model adds the following functionality to HP G60: - Automute of internal speakers - Autoswitch of internal/external mics - Remove SPDIF not physically present BugLink: http://launchpad.net/bugs/587388 Cc: stable@kernel.org Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/HD-Audio-Models.txt | 1 + sound/pci/hda/patch_conexant.c | 57 ++++++++++++++++++++++++++++ 2 files changed, 58 insertions(+) diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ce46fa1e643..37c6aad5e59 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -296,6 +296,7 @@ Conexant 5051 Conexant 5066 ============= laptop Basic Laptop config (default) + hp-laptop HP laptops, e g G60 dell-laptop Dell laptops dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5cdb80edbd7..4f0619908a3 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -116,6 +116,7 @@ struct conexant_spec { unsigned int dell_vostro:1; unsigned int ideapad:1; unsigned int thinkpad:1; + unsigned int hp_laptop:1; unsigned int ext_mic_present; unsigned int recording; @@ -2299,6 +2300,18 @@ static void cxt5066_ideapad_automic(struct hda_codec *codec) } } +/* toggle input of built-in digital mic and mic jack appropriately */ +static void cxt5066_hp_laptop_automic(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_jack_detect(codec, 0x1b); + snd_printdd("CXT5066: external microphone present=%d\n", present); + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 1 : 3); +} + + /* toggle input of built-in digital mic and mic jack appropriately order is: external mic -> dock mic -> interal mic */ static void cxt5066_thinkpad_automic(struct hda_codec *codec) @@ -2407,6 +2420,20 @@ static void cxt5066_ideapad_event(struct hda_codec *codec, unsigned int res) } } +/* unsolicited event for jack sensing */ +static void cxt5066_hp_laptop_event(struct hda_codec *codec, unsigned int res) +{ + snd_printdd("CXT5066_hp_laptop: unsol event %x (%x)\n", res, res >> 26); + switch (res >> 26) { + case CONEXANT_HP_EVENT: + cxt5066_hp_automute(codec); + break; + case CONEXANT_MIC_EVENT: + cxt5066_hp_laptop_automic(codec); + break; + } +} + /* unsolicited event for jack sensing */ static void cxt5066_thinkpad_event(struct hda_codec *codec, unsigned int res) { @@ -2989,6 +3016,14 @@ static struct hda_verb cxt5066_init_verbs_portd_lo[] = { { } /* end */ }; + +static struct hda_verb cxt5066_init_verbs_hp_laptop[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -3004,6 +3039,8 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_ideapad_automic(codec); else if (spec->thinkpad) cxt5066_thinkpad_automic(codec); + else if (spec->hp_laptop) + cxt5066_hp_laptop_automic(codec); } cxt5066_set_mic_boost(codec); return 0; @@ -3031,6 +3068,7 @@ enum { CXT5066_DELL_VOSTO, /* Dell Vostro 1015i */ CXT5066_IDEAPAD, /* Lenovo IdeaPad U150 */ CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ + CXT5066_HP_LAPTOP, /* HP Laptop */ CXT5066_MODELS }; @@ -3041,6 +3079,7 @@ static const char *cxt5066_models[CXT5066_MODELS] = { [CXT5066_DELL_VOSTO] = "dell-vostro", [CXT5066_IDEAPAD] = "ideapad", [CXT5066_THINKPAD] = "thinkpad", + [CXT5066_HP_LAPTOP] = "hp-laptop", }; static struct snd_pci_quirk cxt5066_cfg_tbl[] = { @@ -3052,6 +3091,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), + SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), @@ -3116,6 +3156,23 @@ static int patch_cxt5066(struct hda_codec *codec) spec->num_init_verbs++; spec->dell_automute = 1; break; + case CXT5066_HP_LAPTOP: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_hp_laptop_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_hp_laptop; + spec->num_init_verbs++; + spec->hp_laptop = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + spec->multiout.dig_out_nid = 0; + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; + case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- cgit v1.2.3 From 7b6717e144de6592e614fd7fc3b914b6bf686a9d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 2 Sep 2010 17:13:15 +0800 Subject: ALSA: usb-audio: Assume first control interface is for audio For devices with more than one control interface, let's assume the first one contains the audio controls. Unfortunately, there is no field in any of the descriptors to tell us whether a control interface is for audio or MIDI controls, so a better check is not easy to implement. On a composite device with audio and MIDI functions, for example, the code currently overwrites chip->ctrl_intf, causing operations on the control interface to fail if they are issued after the device probe. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/card.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 9feb00c831a..b443a33d31c 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -465,7 +465,13 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __error; } - chip->ctrl_intf = alts; + /* + * For devices with more than one control interface, we assume the + * first contains the audio controls. We might need a more specific + * check here in the future. + */ + if (!chip->ctrl_intf) + chip->ctrl_intf = alts; if (err > 0) { /* create normal USB audio interfaces */ -- cgit v1.2.3 From a2acad8298a42b7be684a32fafaf83332bba9c2b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 3 Sep 2010 10:53:11 +0200 Subject: ALSA: usb-audio: fix detection of vendor-specific device protocol settings The Audio Class v2 support code in 2.6.35 added checks for the bInterfaceProtocol field. However, there are devices (usually those detected by vendor-specific quirks) that do not have one of the predefined values in this field, which made the driver reject them. To fix this regression, restore the old behaviour, i.e., assume that a device with an unknown bInterfaceProtocol field (other than UAC_VERSION_2) has more or less UAC-v1-compatible descriptors. [compile warning fixes by tiwai] Signed-off-by: Clemens Ladisch Cc: Daniel Mack Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.c | 9 +++++---- sound/usb/clock.c | 3 +-- sound/usb/endpoint.c | 11 ++++++----- sound/usb/format.c | 22 ++++++++++------------ sound/usb/mixer.c | 10 +++++++++- sound/usb/pcm.c | 3 +-- 6 files changed, 32 insertions(+), 26 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index b443a33d31c..32e4be8a187 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -216,6 +216,11 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } switch (protocol) { + default: + snd_printdd(KERN_WARNING "unknown interface protocol %#02x, assuming v1\n", + protocol); + /* fall through */ + case UAC_VERSION_1: { struct uac1_ac_header_descriptor *h1 = control_header; @@ -253,10 +258,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) break; } - - default: - snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); - return -EINVAL; } return 0; diff --git a/sound/usb/clock.c b/sound/usb/clock.c index b853f8df794..7754a103454 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -295,12 +295,11 @@ int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return set_sample_rate_v1(chip, iface, alts, fmt, rate); case UAC_VERSION_2: return set_sample_rate_v2(chip, iface, alts, fmt, rate); } - - return -EINVAL; } diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 1a701f1e8f5..ef0a07e3484 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -275,6 +275,12 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* get audio formats */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d: unknown interface protocol %#02x, assuming v1\n", + dev->devnum, iface_no, altno, protocol); + protocol = UAC_VERSION_1; + /* fall through */ + case UAC_VERSION_1: { struct uac1_as_header_descriptor *as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); @@ -336,11 +342,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno, as->bTerminalLink); continue; } - - default: - snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", - dev->devnum, iface_no, altno, protocol); - continue; } /* get format type */ diff --git a/sound/usb/format.c b/sound/usb/format.c index 3a1375459c0..69148212aa7 100644 --- a/sound/usb/format.c +++ b/sound/usb/format.c @@ -49,7 +49,8 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, u64 pcm_formats; switch (protocol) { - case UAC_VERSION_1: { + case UAC_VERSION_1: + default: { struct uac_format_type_i_discrete_descriptor *fmt = _fmt; sample_width = fmt->bBitResolution; sample_bytes = fmt->bSubframeSize; @@ -64,9 +65,6 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip, format <<= 1; break; } - - default: - return -EINVAL; } pcm_formats = 0; @@ -384,6 +382,10 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, * audio class v2 uses class specific EP0 range requests for that. */ switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: fp->channels = fmt->bNrChannels; ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7); @@ -392,10 +394,6 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, /* fp->channels is already set in this case */ ret = parse_audio_format_rates_v2(chip, fp); break; - default: - snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - return -EINVAL; } if (fp->channels < 1) { @@ -438,6 +436,10 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, fp->channels = 1; switch (protocol) { + default: + snd_printdd(KERN_WARNING "%d:%u:%d : invalid protocol version %d, assuming v1\n", + chip->dev->devnum, fp->iface, fp->altsetting, protocol); + /* fall through */ case UAC_VERSION_1: { struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; brate = le16_to_cpu(fmt->wMaxBitRate); @@ -456,10 +458,6 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, ret = parse_audio_format_rates_v2(chip, fp); break; } - default: - snd_printk(KERN_ERR "%d:%u:%d : invalid protocol version %d\n", - chip->dev->devnum, fp->iface, fp->altsetting, protocol); - return -EINVAL; } return ret; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c166db0057d..3ed3901369c 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -2175,7 +2175,15 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, } host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; - mixer->protocol = get_iface_desc(host_iface)->bInterfaceProtocol; + switch (get_iface_desc(host_iface)->bInterfaceProtocol) { + case UAC_VERSION_1: + default: + mixer->protocol = UAC_VERSION_1; + break; + case UAC_VERSION_2: + mixer->protocol = UAC_VERSION_2; + break; + } if ((err = snd_usb_mixer_controls(mixer)) < 0 || (err = snd_usb_mixer_status_create(mixer)) < 0) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 3634cedf930..3b5135c9306 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -173,13 +173,12 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface, switch (altsd->bInterfaceProtocol) { case UAC_VERSION_1: + default: return init_pitch_v1(chip, iface, alts, fmt); case UAC_VERSION_2: return init_pitch_v2(chip, iface, alts, fmt); } - - return -EINVAL; } /* -- cgit v1.2.3 From 4d155641c81203440da64c4633b4efaab75f63b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 7 Sep 2010 11:58:30 +0200 Subject: ALSA: hda - Add quirk for Lenovo T400s Lenovo T400s requires the quirk to make automatic HP/mic switching working. Reported-by: Frank Becker Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4f0619908a3..71f9d6475b0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3094,6 +3094,7 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400s", CXT5066_THINKPAD), SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD), -- cgit v1.2.3 From 4c25b93223340deff73381cc47f9244fb379a74d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 7 Sep 2010 13:37:10 +0200 Subject: ALSA: virtuoso: work around missing reset in the Xonar DS Windows driver For the WM8776 chip, this driver uses a different sample format and more features than the Windows driver. When rebooting from Linux into Windows, the latter driver does not reset the chip but assumes all its registers have their default settings, so we get garbled sound or, if the output happened to be muted before rebooting, no sound. To make that driver happy, hook our driver's cleanup function into the shutdown notifier and ensure that the chip gets reset. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Nathan Schagen Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen.h | 1 + sound/pci/oxygen/oxygen_lib.c | 21 ++++++++++++++++++--- sound/pci/oxygen/virtuoso.c | 1 + sound/pci/oxygen/xonar_wm87x6.c | 1 + 4 files changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 6147216af74..a3409edcfb5 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -155,6 +155,7 @@ void oxygen_pci_remove(struct pci_dev *pci); int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); int oxygen_pci_resume(struct pci_dev *pci); #endif +void oxygen_pci_shutdown(struct pci_dev *pci); /* oxygen_mixer.c */ diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index fad03d64e3a..7e93cf88443 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -519,16 +519,21 @@ static void oxygen_init(struct oxygen *chip) } } -static void oxygen_card_free(struct snd_card *card) +static void oxygen_shutdown(struct oxygen *chip) { - struct oxygen *chip = card->private_data; - spin_lock_irq(&chip->reg_lock); chip->interrupt_mask = 0; chip->pcm_running = 0; oxygen_write16(chip, OXYGEN_DMA_STATUS, 0); oxygen_write16(chip, OXYGEN_INTERRUPT_MASK, 0); spin_unlock_irq(&chip->reg_lock); +} + +static void oxygen_card_free(struct snd_card *card) +{ + struct oxygen *chip = card->private_data; + + oxygen_shutdown(chip); if (chip->irq >= 0) free_irq(chip->irq, chip); flush_scheduled_work(); @@ -778,3 +783,13 @@ int oxygen_pci_resume(struct pci_dev *pci) } EXPORT_SYMBOL(oxygen_pci_resume); #endif /* CONFIG_PM */ + +void oxygen_pci_shutdown(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct oxygen *chip = card->private_data; + + oxygen_shutdown(chip); + chip->model.cleanup(chip); +} +EXPORT_SYMBOL(oxygen_pci_shutdown); diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index f03a2f2cffe..06c863e86e3 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -95,6 +95,7 @@ static struct pci_driver xonar_driver = { .suspend = oxygen_pci_suspend, .resume = oxygen_pci_resume, #endif + .shutdown = oxygen_pci_shutdown, }; static int __init alsa_card_xonar_init(void) diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index dbc4b89d74e..0b89932fb8c 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -193,6 +193,7 @@ static void xonar_ds_init(struct oxygen *chip) static void xonar_ds_cleanup(struct oxygen *chip) { xonar_disable_output(chip); + wm8776_write(chip, WM8776_RESET, 0); } static void xonar_ds_suspend(struct oxygen *chip) -- cgit v1.2.3 From fe6ce80ae25953d95ebaf9bce27b585218cda25c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 7 Sep 2010 13:38:49 +0200 Subject: ALSA: virtuoso: fix setting of Xonar DS line-in/mic-in controls The Line and Mic inputs cannot be used at the same time, so the driver has to automatically disable one of them if both are set. However, it forgot to notify userspace about this change, so the mixer state would be inconsistent. To fix this, check if the other control gets muted, and send a notification event in this case. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Nathan Schagen Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_wm87x6.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 0b89932fb8c..b82c1cfa96f 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -53,6 +53,8 @@ struct xonar_wm87x6 { struct xonar_generic generic; u16 wm8776_regs[0x17]; u16 wm8766_regs[0x10]; + struct snd_kcontrol *line_adcmux_control; + struct snd_kcontrol *mic_adcmux_control; struct snd_kcontrol *lc_controls[13]; }; @@ -604,6 +606,7 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl, { struct oxygen *chip = ctl->private_data; struct xonar_wm87x6 *data = chip->model_data; + struct snd_kcontrol *other_ctl; unsigned int mux_bit = ctl->private_value; u16 reg; int changed; @@ -611,8 +614,18 @@ static int wm8776_input_mux_put(struct snd_kcontrol *ctl, mutex_lock(&chip->mutex); reg = data->wm8776_regs[WM8776_ADCMUX]; if (value->value.integer.value[0]) { - reg &= ~0x003; reg |= mux_bit; + /* line-in and mic-in are exclusive */ + mux_bit ^= 3; + if (reg & mux_bit) { + reg &= ~mux_bit; + if (mux_bit == 1) + other_ctl = data->line_adcmux_control; + else + other_ctl = data->mic_adcmux_control; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_VALUE, + &other_ctl->id); + } } else reg &= ~mux_bit; changed = reg != data->wm8776_regs[WM8776_ADCMUX]; @@ -964,7 +977,13 @@ static int xonar_ds_mixer_init(struct oxygen *chip) err = snd_ctl_add(chip->card, ctl); if (err < 0) return err; + if (!strcmp(ctl->id.name, "Line Capture Switch")) + data->line_adcmux_control = ctl; + else if (!strcmp(ctl->id.name, "Mic Capture Switch")) + data->mic_adcmux_control = ctl; } + if (!data->line_adcmux_control || !data->mic_adcmux_control) + return -ENXIO; BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { ctl = snd_ctl_new1(&lc_controls[i], chip); -- cgit v1.2.3 From 76195fb096ca6db2f8bbaffb96e3025aaf1649a0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 08:27:02 +0200 Subject: ALSA: usb - Release capture substream URBs properly Due to the wrong "return" in the loop, a capture substream won't be released at disconnection properly if the device is capture only and has no playback substream. This caused Oops occasionally at the device reconnection. Reported-by: Kim Minhyoung Cc: Signed-off-by: Takashi Iwai --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 32e4be8a187..4eabafa5b03 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -126,7 +126,7 @@ static void snd_usb_stream_disconnect(struct list_head *head) for (idx = 0; idx < 2; idx++) { subs = &as->substream[idx]; if (!subs->num_formats) - return; + continue; snd_usb_release_substream_urbs(subs, 1); subs->interface = -1; } -- cgit v1.2.3 From a769cbcf60cee51f4431c0938acd39e7e5b76b8d Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 7 Sep 2010 14:36:22 -0500 Subject: ALSA: hda - Add errata initverb sequence for CS42xx codecs Add init verb sequence for errata ER880C3 http://www.cirrus.com/en/pubs/errata/ER880C3.pdf Signed-off-by: Brian Austin Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_cirrus.c | 50 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4ef5efaaaef..488fd9ade1b 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -972,6 +972,53 @@ static struct hda_verb cs_coef_init_verbs[] = { {} /* terminator */ }; +/* Errata: CS4207 rev C0/C1/C2 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ER880C3.pdf + * + * 6. At high temperature (TA > +85°C), the digital supply current (IVD) + * may be excessive (up to an additional 200 μA), which is most easily + * observed while the part is being held in reset (RESET# active low). + * + * Root Cause: At initial powerup of the device, the logic that drives + * the clock and write enable to the S/PDIF SRC RAMs is not properly + * initialized. + * Certain random patterns will cause a steady leakage current in those + * RAM cells. The issue will resolve once the SRCs are used (turned on). + * + * Workaround: The following verb sequence briefly turns on the S/PDIF SRC + * blocks, which will alleviate the issue. + */ + +static struct hda_verb cs_errata_init_verbs[] = { + {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ + {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x11, AC_VERB_SET_COEF_INDEX, 0x0008}, + {0x11, AC_VERB_SET_PROC_COEF, 0x9999}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0017}, + {0x11, AC_VERB_SET_PROC_COEF, 0xa412}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0001}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0009}, + + {0x07, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Rx: D0 */ + {0x08, AC_VERB_SET_POWER_STATE, 0x00}, /* S/PDIF Tx: D0 */ + + {0x11, AC_VERB_SET_COEF_INDEX, 0x0017}, + {0x11, AC_VERB_SET_PROC_COEF, 0x2412}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0008}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0000}, + {0x11, AC_VERB_SET_COEF_INDEX, 0x0001}, + {0x11, AC_VERB_SET_PROC_COEF, 0x0008}, + {0x11, AC_VERB_SET_PROC_STATE, 0x00}, + + {0x07, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Rx: D3 */ + {0x08, AC_VERB_SET_POWER_STATE, 0x03}, /* S/PDIF Tx: D3 */ + /*{0x01, AC_VERB_SET_POWER_STATE, 0x03},*/ /* AFG: D3 This is already handled */ + + {} /* terminator */ +}; + /* SPDIF setup */ static void init_digital(struct hda_codec *codec) { @@ -991,6 +1038,9 @@ static int cs_init(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; + /* init_verb sequence for C0/C1/C2 errata*/ + snd_hda_sequence_write(codec, cs_errata_init_verbs); + snd_hda_sequence_write(codec, cs_coef_init_verbs); if (spec->gpio_mask) { -- cgit v1.2.3 From 080dc7bc2562615a5be0a705a9d1a8c24eb198d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 08:38:41 +0200 Subject: ALSA: hda - Enable PC-beep for EeePC with ALC269 codec EeePC 1001HAG has a similar problem like other ASUS machine, which doesn't set the codec SSID properly for indicating the beep capability. To enable PC-beep again, put this to the whitelist. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 627bf996336..bcbf9160ed8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5334,6 +5334,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, static struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), + SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; -- cgit v1.2.3 From e4ee8dd8afcbcbe502fa8a3d3af6eb09c96dd806 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 09:58:12 +0200 Subject: ALSA: msnd-classic: Fix invalid cfg parameter The driver doesn't probe the device properly because of left-over cfg[] that isn't used at all for msnd-classic device. This is only for msnd- pinnacle. Signed-off-by: Takashi Iwai --- sound/isa/msnd/msnd_pinnacle.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c index 5f3e68401f9..91d6023a63e 100644 --- a/sound/isa/msnd/msnd_pinnacle.c +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -764,9 +764,9 @@ static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static long mem[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +#ifndef MSND_CLASSIC static long cfg[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; -#ifndef MSND_CLASSIC /* Extra Peripheral Configuration (Default: Disable) */ static long ide_io0[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static long ide_io1[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; @@ -894,7 +894,11 @@ static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) struct snd_card *card; struct snd_msnd *chip; - if (has_isapnp(idx) || cfg[idx] == SNDRV_AUTO_PORT) { + if (has_isapnp(idx) +#ifndef MSND_CLASSIC + || cfg[idx] == SNDRV_AUTO_PORT +#endif + ) { printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); return -ENODEV; } -- cgit v1.2.3 From 27f7ad53829f79e799a253285318bff79ece15bd Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 6 Sep 2010 09:13:45 +0200 Subject: ALSA: seq/oss - Fix double-free at error path of snd_seq_oss_open() The error handling in snd_seq_oss_open() has several bad codes that do dereferecing released pointers and double-free of kmalloc'ed data. The object dp is release in free_devinfo() that is called via private_free callback. The rest shouldn't touch this object any more. The patch changes delete_port() to call kfree() in any case, and gets rid of unnecessary calls of destructors in snd_seq_oss_open(). Fixes CVE-2010-3080. Reported-and-tested-by: Tavis Ormandy Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_init.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index 685712276ac..69cd7b3c362 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -281,13 +281,10 @@ snd_seq_oss_open(struct file *file, int level) return 0; _error: - snd_seq_oss_writeq_delete(dp->writeq); - snd_seq_oss_readq_delete(dp->readq); snd_seq_oss_synth_cleanup(dp); snd_seq_oss_midi_cleanup(dp); - delete_port(dp); delete_seq_queue(dp->queue); - kfree(dp); + delete_port(dp); return rc; } @@ -350,8 +347,10 @@ create_port(struct seq_oss_devinfo *dp) static int delete_port(struct seq_oss_devinfo *dp) { - if (dp->port < 0) + if (dp->port < 0) { + kfree(dp); return 0; + } debug_printk(("delete_port %i\n", dp->port)); return snd_seq_event_port_detach(dp->cseq, dp->port); -- cgit v1.2.3 From 122661b67899980f1372812d907e73ebcfb3d037 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 8 Sep 2010 14:57:04 +0200 Subject: ALSA: hda - Fix wrong HP pin detection in snd_hda_parse_pin_def_config() snd_hda_parse_pin_def_config() has some workaround for re-assigning some pins declared as headphones to line-outs. This didn't work properly for some cases because it used memmove() stupidly wrongly. Reference: Novell bnc#637263 https://bugzilla.novell.com/show_bug.cgi?id=637263 Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3827092cc1d..14829210ef0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4536,7 +4536,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_outs--; memmove(cfg->hp_pins + i, cfg->hp_pins + i + 1, sizeof(cfg->hp_pins[0]) * (cfg->hp_outs - i)); - memmove(sequences_hp + i - 1, sequences_hp + i, + memmove(sequences_hp + i, sequences_hp + i + 1, sizeof(sequences_hp[0]) * (cfg->hp_outs - i)); } } -- cgit v1.2.3 From a7a13d0676335a7dc9dd72264cca02606e43aaba Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 9 Sep 2010 00:11:41 +0200 Subject: ALSA: rawmidi: fix the get next midi device ioctl If we pass in a device which is higher than SNDRV_RAWMIDI_DEVICES then the "next device" should be -1. This function just returns device + 1. But the main thing is that "device + 1" can lead to a (harmless) integer overflow and that annoys static analysis tools. [fix the case for device == SNDRV_RAWMIDI_DEVICE by tiwai] Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/core/rawmidi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index eb68326c37d..a7868ad4d53 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -829,6 +829,8 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, if (get_user(device, (int __user *)argp)) return -EFAULT; + if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */ + device = SNDRV_RAWMIDI_DEVICES - 1; mutex_lock(®ister_mutex); device = device < 0 ? 0 : device + 1; while (device < SNDRV_RAWMIDI_DEVICES) { -- cgit v1.2.3