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[alsa-devel] [PATCH 2/31] HDA patch_via.c: Change get_codec_type argument to hda_codec type
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Change get_codec_type argument to hda_codec type Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-04 23:01:14.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-04 23:05:10.000000000 +0800 @@ -88,8 +88,9 @@ CODEC_TYPES, }; -static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) +static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { + u32 vendor_id = codec->vendor_id; u16 ven_id = vendor_id >> 16; u16 dev_id = vendor_id & 0xffff; enum VIA_HDA_CODEC codec_type; @@ -141,7 +142,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { if (size < 4 * sizeof(unsigned int)) return -ENOMEM; @@ -163,7 +164,7 @@ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); - if (get_codec_type(codec->vendor_id) == VT1708S + if (get_codec_type(codec) == VT1708S && (nid == 0x1a || nid == 0x1e)) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2;

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[alsa-devel] [PATCH 13/31] HDA patch_via.c: Refresh front playback mute if hp plug-in mutes speaker.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Refresh front playback mute if hp plug-in mutes speaker. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:13.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:22.000000000 +0800 @@ -1346,14 +1346,26 @@ /* mute internal speaker if HP is plugged */ static void via_hp_automute(struct hda_codec *codec) { - unsigned int present; + unsigned int present = 0; struct via_spec *spec = codec->spec; + char name[32] = "Front Playback Switch"; present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0], - HDA_OUTPUT, 0, HDA_AMP_MUTE, - present ? HDA_AMP_MUTE : 0); + + if (!spec->hp_independent_mode) { + struct snd_card *card = codec->bus->card; + struct snd_kcontrol *kctl; + /* auto mute */ + snd_hda_codec_amp_stereo( + codec, spec->autocfg.line_out_pins[0], HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + /* notify change */ + list_for_each_entry(kctl, &card->controls, list) + if (strcmp(kctl->id.name, name) == 0) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &kctl->id); + } } static void via_gpio_control(struct hda_codec *codec)

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[alsa-devel] [PATCH 17/31] HDA patch_via.c: Add second S/PDIF out control for VT1708S and VT1702.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add second S/PDIF out control for VT1708S and VT1702. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:59.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:11:02.000000000 +0800 @@ -1265,6 +1265,13 @@ if (err < 0) return err; spec->multiout.share_spdif = 1; + /* lydia add for second spdif out */ + if (spec->slave_dig_outs[0]) { + err = snd_hda_create_spdif_out_ctls(codec, + spec->slave_dig_outs[0]); + if (err < 0) + return err; + } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -3127,7 +3134,7 @@ }; static struct hda_pcm_stream vt1708S_pcm_digital_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 2, /* NID is set in via_build_pcms */

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Re: [alsa-devel] [PATCH 18/31] HDA patch_via.c: Add 48K sample rate limit for S/PDIF streams.
by Demian Martin 08 Oct '09

08 Oct '09

Li Bo: I'm not sure how changing the sample rate for SPDIF would have an effect of the SNR if you are only passing digital signals. I have had excellent measured results with the Via chipsets at all sample rates up to 192 KHz at 24 bits on the digital output. Perhaps you can explain how you measured the performance and that may explain your results. Normally if all of the data of the original file is present then any limit on SNR is a function of the DAC. If the data is not the same as the original file then there are more serious issues to be resolved, and it probably would not be useable with Dolby Digital or DTS. Higher sample rates for digital content and digital interfaces are becoming more common and I think it would be a serious mistake to limit the performance of the platform. >Hi, Takashi >According to our HW test, S/PDIF at 48K sample rate has best SNR, so this limit can make >S/PDIF work better. Demian Martin Product Design Service 784 Cary Drive San Leandro, CA 94577 209 613 6990

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[alsa-devel] [PATCH 8/31] HDA patch_via.c: Change VT1708S & VT1702 hp mode controls
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Change VT1708S & VT1702 hp mode controls For VT1708S and VT1702, deactivate "Headphone Playback Volume" and "Headphone Playback Mute" control if "Independent HP" mode is OFF. and rename VT1702 "Independent HP" text. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:09:18.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:09:33.000000000 +0800 @@ -566,9 +566,26 @@ unsigned int pinsel = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0x00); + static int hp_ctl_initialized = -1; ucontrol->value.enumerated.item[0] = pinsel; - + /* update HP controls on first time */ + if ((spec->codec_type == VT1708S + || spec->codec_type == VT1702) + && hp_ctl_initialized == -1) { + struct snd_card *card = codec->bus->card; + struct snd_kcontrol *kctl; + /* deactive hp volume & switch in independent mode */ + list_for_each_entry(kctl, &card->controls, list) + if (spec->hp_independent_mode + && (strcmp(kctl->id.name, + "Headphone Playback Volume") == 0 + || strcmp(kctl->id.name, + "Headphone Playback Switch") == 0)) + kctl->vd[0].access |= + SNDRV_CTL_ELEM_ACCESS_INACTIVE; + hp_ctl_initialized = 1; + } return 0; } @@ -3342,11 +3359,13 @@ static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { - int err; - + int err, i; + struct hda_input_mux *imux; + static const char *texts[] = { "ON (need reboot)", + "OFF (need reboot)", + NULL}; if (!pin) return 0; - spec->multiout.hp_nid = 0x1D; err = via_add_control(spec, VIA_CTL_WIDGET_VOL, @@ -3361,8 +3380,18 @@ if (err < 0) return err; - create_hp_imux(spec); + imux = &spec->private_imux[1]; + /* for hp mode select */ + i = 0; + while (texts[i] != NULL) { + imux->items[imux->num_items].label = texts[i]; + imux->items[imux->num_items].index = i; + imux->num_items++; + i++; + } + + spec->hp_mux = &spec->private_imux[1]; return 0; }

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[alsa-devel] [PATCH 14/31] HDA patch_via.c: Add Jack detect feature for VT1708.
by Li Bo 08 Oct '09

08 Oct '09

[ALSA] HDA VIA: Add Jack detect feature for VT1708. Signed-off-by: Lydia Wang <lydiawang(a)viatech.com.cn> Index: sound-2.6/sound/pci/hda/patch_via.c =================================================================== --- sound-2.6.orig/sound/pci/hda/patch_via.c 2009-10-05 15:10:22.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_via.c 2009-10-05 15:10:36.000000000 +0800 @@ -257,6 +257,11 @@ unsigned int smart51_enabled; enum VIA_HDA_CODEC codec_type; + /* work to check hp jack state */ + struct hda_codec *codec; + struct delayed_work hp_jack_work; + int vt1708_jack_detectect; + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -966,6 +971,8 @@ {0x20, AC_VERB_SET_CONNECT_SEL, 0x1}, /* PW9 Output enable */ {0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* power down jack detect function */ + {0x1, 0xf81, 0x1}, { } }; @@ -1340,6 +1347,10 @@ return; via_free_kctls(codec); + if (spec->codec_type == VT1708) { + cancel_delayed_work(&spec->hp_jack_work); + flush_scheduled_work(); + } kfree(codec->spec); } @@ -1724,6 +1735,52 @@ return; } +static int vt1708_jack_detectect_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = + !((snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8) & 0x1); + ucontrol->value.integer.value[0] = spec->vt1708_jack_detectect; + return 0; +} + +static int vt1708_jack_detectect_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + int change; + + if (spec->codec_type != VT1708) + return 0; + spec->vt1708_jack_detectect = ucontrol->value.integer.value[0]; + snd_hda_codec_write(codec, 0x1, 0, 0xf81, !spec->vt1708_jack_detectect); + change = (0x1 & (snd_hda_codec_read(codec, 0x1, 0, 0xf84, 0) >> 8)) + == !spec->vt1708_jack_detectect; + if (spec->vt1708_jack_detectect) { + mute_aa_path(codec, 1); + notify_aa_path_ctls(codec); + } + return change; +} + +static struct snd_kcontrol_new vt1708_jack_detectect[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Jack Detect", + .count = 1, + .info = snd_ctl_boolean_mono_info, + .get = vt1708_jack_detectect_get, + .put = vt1708_jack_detectect_put, + }, + {} /* end */ +}; + static int vt1708_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1751,6 +1808,10 @@ err = vt1708_auto_create_analog_input_ctls(spec, &spec->autocfg); if (err < 0) return err; + /* add jack detect on/off control */ + err = snd_hda_add_new_ctls(codec, vt1708_jack_detectect); + if (err < 0) + return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; @@ -1784,6 +1845,24 @@ return 0; } +static void update_hp_jack_state(struct work_struct *work) +{ + struct via_spec *spec = container_of(work, struct via_spec, + hp_jack_work.work); + static int present = 1; + + if (spec->codec_type != VT1708) + return; + /* if jack state toggled */ + if (present + != (snd_hda_codec_read(spec->codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) >> 31)) { + present ^= 1; + via_hp_automute(spec->codec); + } + schedule_delayed_work(&spec->hp_jack_work, msecs_to_jiffies(100)); +} + static int get_mux_nids(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1860,7 +1939,9 @@ #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - + spec->codec = codec; + INIT_DELAYED_WORK(&spec->hp_jack_work, update_hp_jack_state); + schedule_delayed_work(&spec->hp_jack_work, msecs_to_jiffies(100)); return 0; }

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Re: [alsa-devel] [PATCH REALTEK] Added quirk to enable sound on Toshiba NB200
by Takashi Iwai 08 Oct '09

08 Oct '09

At Wed, 23 Sep 2009 00:39:45 -0500 (CDT), Manoj Iyer wrote: > > > >From c96629e96fecbe59de7124ec913019ed378d0029 Mon Sep 17 00:00:00 2001 > From: Manoj Iyer <manoj.iyer(a)canonical.com> > Date: Tue, 22 Sep 2009 18:33:29 -0500 > Subject: [PATCH] [PATCH REALTEK] Added quirk to enable sound on Toshiba NB200 > > Patch was tested on Toshiba NB200 and is found to enable sound. So, it doesn't work without the model quirk? Could you give alsa-info.sh output (run with --no-upload option)? thanks, Takashi > > Signed-off-by: Manoj Iyer <manoj.iyer(a)canonical.com> > Cc: stable(a)kernel.org > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 files changed, 1 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 30eeb30..713bf69 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -16876,6 +16876,7 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { > SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), > SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", > ALC662_3ST_6ch_DIG), > + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), > SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), > SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", > ALC662_3ST_6ch_DIG), > -- > 1.6.3.3 > > > --- manjo >

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[alsa-devel] [PATCHv3 1/7] OMAP4: PMIC: Add support for twl6030 codec
by Lopez Cruz, Misael 08 Oct '09

08 Oct '09

In order to have TWL6030 CODEC driver as a platform driver, codec data should be passed through twl_platform_data structure. For twl6030 audio codec, the following data may be passed: - audpwron_gpio: gpio line used to power-up/down the codec. A low-to-high transition powers codec up. Setting audpwron_gpio to a negative value means that codec will use manual power sequence instead of automatic sequence - naudint_irq: irq line for audio interrupt. twl6030 drives NAUDINT line to low when an interrupt (codec ready, plug insertion/removal, etc) is detected However, codec driver can operate if any or none of them are passed. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- drivers/mfd/twl-core.c | 15 +++++++++++++++ include/linux/i2c/twl.h | 7 +++++++ 2 files changed, 22 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/twl-core.c b/drivers/mfd/twl-core.c index af4cf47..6432af1 100644 --- a/drivers/mfd/twl-core.c +++ b/drivers/mfd/twl-core.c @@ -108,6 +108,12 @@ #define twl_has_mmc() false #endif +#if defined(CONFIG_SND_SOC_TWL6030) || defined(CONFIG_SND_SOC_TWL6030_MODULE) +#define twl_has_codec() true +#else +#define twl_has_codec() false +#endif + #if defined(CONFIG_TWL4030_USB) || defined(CONFIG_TWL4030_USB_MODULE) #define twl_has_usb() true #else @@ -190,6 +196,7 @@ #define BCI_SUB_CHIP_ID SUB_CHIP_ID1 #define GPIO_SUB_CHIP_ID 0 /* NOT SUPPORTED IN TWL6030 */ #define KEYPAD_SUB_CHIP_ID 0 /* ADDED FOR COMPILATION ONLY */ +#define CODEC_SUB_CHIP_ID SUB_CHIP_ID3 /* subchip/slave 0 0x48 - POWER */ #define TWL6030_BASEADD_RTC 0x0000 @@ -632,6 +639,14 @@ add_children(struct twl_platform_data *pdata, unsigned long features) if (IS_ERR(child)) return PTR_ERR(child); } + + if (twl_has_codec()) { + child = add_child(CODEC_SUB_CHIP_ID, "twl6030_codec", + pdata->codec, sizeof(*pdata->codec), false, + 0, 0); + if (IS_ERR(child)) + return PTR_ERR(child); + } #endif if (twl_has_usb() && pdata->usb) { diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h index b687a8b..e76ca9b 100644 --- a/include/linux/i2c/twl.h +++ b/include/linux/i2c/twl.h @@ -472,6 +472,12 @@ struct twl_usb_data { enum twl_usb_mode usb_mode; }; +struct twl_codec_data { + /* twl6030 */ + int audpwron_gpio; /* audio power-on gpio */ + int naudint_irq; /* audio interrupt */ +}; + struct twl_platform_data { unsigned irq_base, irq_end; struct twl_bci_platform_data *bci; @@ -479,6 +485,7 @@ struct twl_platform_data { struct twl_madc_platform_data *madc; struct twl_keypad_data *keypad; struct twl_usb_data *usb; + struct twl_codec_data *codec; /* LDO regulators common to TWL4030/TWL6030 */ struct regulator_init_data *vdac; -- 1.5.4.3

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[alsa-devel] Possibility to build alsa driver without having exact kernel source?
by Christian Steffen 07 Oct '09

07 Oct '09

Hello list, I need a new driver for a atom cpu driven device. It needs the "snd_intel_hda" driver to work as it has intels Poulsbo chipset. Currently I have a compiled module, but sound has a constant clicking noise with this one. Therefore I'd like to try an updated alsa driver, which hopefully fixes a problem on Poulsbo chipsets with stepping D0 or newer. My problem is that I only have a compiled running kernel, but no source from it. All I can see (trough uname -a) is: Linux joggler 2.6.30 #4 SMP Tue Jun 23 21:55:08 BST 2009 i686 GNU/Linux So I know, the kernel is a 2.6.30, arch i686. I tried to compile the driver now just with the standard kernel 2.6.30 source from kernel. org. But when trying to load these modules, I get "Invalid module format". Is there any chance to get a working compiled driver without having more information? Thanks! Christian Steffen P Bitte drucken Sie diese Mail nur, wenn dies notwendig ist. please don't print this e-mail unless you really need to

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Re: [alsa-devel] Patch for Zotac IONITX model detection
by gus.alsa＠bourg.net 07 Oct '09

07 Oct '09

Aha, sorry - I misunderstood what was being asked to test. Two channel audio is working correctly with the development snapshot. Thanks, Gus On Wed 7/10/09 10:16 AM , Takashi Iwai <tiwai(a)suse.de> wrote: > At Wed, 7 Oct 2009 03:12:49 +0000, > wrote: > > > > Hello Takashi Iwai, > > > > I tested your latest snapshot and it did not detect the 6 channel > audio. > This is so per design. The default auto-parser doesn't enable > 6-channel since otherwise you get problems with the automatic HP / > mic > jack detection. > Does it work except for 6-channel things? > thanks, > Takashi > _______________________________________________ > Alsa-devel mailing list > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > Message sent via Atmail Open - http://atmail.org/

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