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[alsa-devel] [PATCH] ALSA: hda - add quirks for some 82801H variants to use ALC883_MITAC
by Luke Yelavich 16 Jan '09

16 Jan '09

Add the 82801H variants 1071:8227 and 8086:2503 to use ALC883_MITAC Reference: Ubuntu bug 210865 https://bugs.launchpad.net/bugs/210865 Signed-off-by: Luke Yelavich <themuso(a)ubuntu.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 82dd084..5d249a5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8478,6 +8478,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), @@ -8526,6 +8527,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} -- 1.6.0.4

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[alsa-devel] [PATCH 1/1] ALSA: ASoC: DaVinci: Fix SFFSDR compilation error.
by Hugo Villeneuve 15 Jan '09

15 Jan '09

Remove dependency on sffsdr_fpga_set_codec_fs() when the SFFSDR FPGA module is not selected. Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> --- sound/soc/davinci/davinci-sffsdr.c | 20 +++++++++++++++++--- 1 files changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 3e8e6cd..e3ea07a 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -25,7 +25,9 @@ #include <asm/gpio.h> #include <asm/dma.h> #include <asm/mach-types.h> +#ifdef CONFIG_SFFSDR_FPGA #include <asm/plat-sffsdr/sffsdr-fpga.h> +#endif #include <mach/asp.h> #include <mach/edma.h> @@ -42,6 +44,17 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, int fs; int ret = 0; + /* Fsref can be 32000, 44100 or 48000. */ + fs = params_rate(params); + +#ifndef CONFIG_SFFSDR_FPGA + /* Without the FPGA module, the Fs is fixed at 44100 Hz */ + if (fs != 44100) { + pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n"); + return -EINVAL; + } +#endif + /* Set cpu DAI configuration: * CLKX and CLKR are the inputs for the Sample Rate Generator. * FSX and FSR are outputs, driven by the sample Rate Generator. */ @@ -52,12 +65,13 @@ static int sffsdr_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Fsref can be 32000, 44100 or 48000. */ - fs = params_rate(params); - pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs); +#ifndef CONFIG_SFFSDR_FPGA + return 0; +#else return sffsdr_fpga_set_codec_fs(fs); +#endif } static struct snd_soc_ops sffsdr_ops = { -- 1.5.4.5

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[alsa-devel] [PATCH 1/1] ALSA: ASoC: DaVinci: Fix SFFSDR init bug.
by Hugo Villeneuve 15 Jan '09

15 Jan '09

Removed call to machine_is_sffsdr() in module init function because it always returns zero and causes GCC to optimise and remove all functions of the module. Signed-off-by: Hugo Villeneuve <hugo(a)hugovil.com> --- sound/soc/davinci/davinci-sffsdr.c | 3 --- 1 files changed, 0 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c index 1bbde3e..3e8e6cd 100644 --- a/sound/soc/davinci/davinci-sffsdr.c +++ b/sound/soc/davinci/davinci-sffsdr.c @@ -115,9 +115,6 @@ static int __init sffsdr_init(void) { int ret; - if (!machine_is_sffsdr()) - return -EINVAL; - sffsdr_snd_device = platform_device_alloc("soc-audio", 0); if (!sffsdr_snd_device) { printk(KERN_ERR "platform device allocation failed\n"); -- 1.5.4.5

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[alsa-devel] [PATCH] snd-aoa: handle older machines
by Johannes Berg 15 Jan '09

15 Jan '09

This patch changes snd-aoa to handle some older machines that are currently handled by snd-powermac. snd-aoa has a number of advantages though, notably it can autoload better and is generally a more modern driver. By hardcoding the accepted device-ids (last hunk of the patch) I'm trying to avoid regressions because this driver will otherwise load automatically and not let snd-powermac load. People who are unhappy with snd-powermac and have a device-id property in the device tree are encouraged to read this patch and make a patch to amend this as appropriate. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- sound/aoa/fabrics/layout.c | 74 ++++++++++++++++++++++++++++++--------- sound/aoa/soundbus/i2sbus/core.c | 22 ++++++++--- 2 files changed, 74 insertions(+), 22 deletions(-) --- wireless-testing.orig/sound/aoa/fabrics/layout.c 2009-01-13 17:01:04.000000000 +0100 +++ wireless-testing/sound/aoa/fabrics/layout.c 2009-01-15 18:15:28.000000000 +0100 @@ -1,16 +1,14 @@ /* - * Apple Onboard Audio driver -- layout fabric + * Apple Onboard Audio driver -- layout/machine id fabric * - * Copyright 2006 Johannes Berg <johannes(a)sipsolutions.net> + * Copyright 2006-2008 Johannes Berg <johannes(a)sipsolutions.net> * * GPL v2, can be found in COPYING. * * - * This fabric module looks for sound codecs - * based on the layout-id property in the device tree. - * + * This fabric module looks for sound codecs based on the + * layout-id or device-id property in the device tree. */ - #include <asm/prom.h> #include <linux/list.h> #include <linux/module.h> @@ -63,7 +61,7 @@ struct codec_connect_info { #define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF (1<<0) struct layout { - unsigned int layout_id; + unsigned int layout_id, device_id; struct codec_connect_info codecs[MAX_CODECS_PER_BUS]; int flags; @@ -111,6 +109,10 @@ MODULE_ALIAS("sound-layout-96"); MODULE_ALIAS("sound-layout-98"); MODULE_ALIAS("sound-layout-100"); +MODULE_ALIAS("aoa-device-id-14"); +MODULE_ALIAS("aoa-device-id-22"); +MODULE_ALIAS("aoa-device-id-35"); + /* onyx with all but microphone connected */ static struct codec_connection onyx_connections_nomic[] = { { @@ -518,6 +520,27 @@ static struct layout layouts[] = { .connections = onyx_connections_noheadphones, }, }, + /* PowerMac3,4 */ + { .device_id = 14, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + /* PowerMac3,6 */ + { .device_id = 22, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + /* PowerBook5,2 */ + { .device_id = 35, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, {} }; @@ -526,7 +549,7 @@ static struct layout *find_layout_by_id( struct layout *l; l = layouts; - while (l->layout_id) { + while (l->codecs[0].name) { if (l->layout_id == id) return l; l++; @@ -534,6 +557,19 @@ static struct layout *find_layout_by_id( return NULL; } +static struct layout *find_layout_by_device(unsigned int id) +{ + struct layout *l; + + l = layouts; + while (l->codecs[0].name) { + if (l->device_id == id) + return l; + l++; + } + return NULL; +} + static void use_layout(struct layout *l) { int i; @@ -938,8 +974,8 @@ static struct aoa_fabric layout_fabric = static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) { struct device_node *sound = NULL; - const unsigned int *layout_id; - struct layout *layout; + const unsigned int *id; + struct layout *layout = NULL; struct layout_dev *ldev = NULL; int err; @@ -952,15 +988,18 @@ static int aoa_fabric_layout_probe(struc if (sound->type && strcasecmp(sound->type, "soundchip") == 0) break; } - if (!sound) return -ENODEV; + if (!sound) + return -ENODEV; - layout_id = of_get_property(sound, "layout-id", NULL); - if (!layout_id) - goto outnodev; - printk(KERN_INFO "snd-aoa-fabric-layout: found bus with layout %d\n", - *layout_id); + id = of_get_property(sound, "layout-id", NULL); + if (id) { + layout = find_layout_by_id(*id); + } else { + id = of_get_property(sound, "device-id", NULL); + if (id) + layout = find_layout_by_device(*id); + } - layout = find_layout_by_id(*layout_id); if (!layout) { printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n"); goto outnodev; @@ -976,6 +1015,7 @@ static int aoa_fabric_layout_probe(struc ldev->layout = layout; ldev->gpio.node = sound->parent; switch (layout->layout_id) { + case 0: /* anything with device_id, not layout_id */ case 41: /* that unknown machine no one seems to have */ case 51: /* PowerBook5,4 */ case 58: /* Mac Mini */ --- wireless-testing.orig/sound/aoa/soundbus/i2sbus/core.c 2009-01-13 17:01:04.000000000 +0100 +++ wireless-testing/sound/aoa/soundbus/i2sbus/core.c 2009-01-15 18:15:34.000000000 +0100 @@ -1,7 +1,7 @@ /* * i2sbus driver * - * Copyright 2006 Johannes Berg <johannes(a)sipsolutions.net> + * Copyright 2006-2008 Johannes Berg <johannes(a)sipsolutions.net> * * GPL v2, can be found in COPYING. */ @@ -186,13 +186,25 @@ static int i2sbus_add_dev(struct macio_d } } if (i == 1) { - const u32 *layout_id = - of_get_property(sound, "layout-id", NULL); - if (layout_id) { - layout = *layout_id; + const u32 *id = of_get_property(sound, "layout-id", NULL); + + if (id) { + layout = *id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); ok = 1; + } else { + id = of_get_property(sound, "device-id", NULL); + /* + * We probably cannot handle all device-id machines, + * so restrict to those we do handle for now. + */ + if (id && (*id == 22 || *id == 14 || *id == 35)) { + snprintf(dev->sound.modalias, 32, + "aoa-device-id-%d", *id); + ok = 1; + layout = -1; + } } } /* for the time being, until we can handle non-layout-id

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[alsa-devel] [PATCH] snd-aoa: handle master-amp if present
by Johannes Berg 15 Jan '09

15 Jan '09

Some machines have a master amp GPIO that needs to be toggled to get sound output, in addition to speaker/headphone/line-out amps. This makes snd-aoa handle it, if present in the device tree, thus making snd-aoa be able to output sound on PowerMac3,6, which was previously handled by snd-powermac which also doesn't use the master amp GPIO. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- sound/aoa/aoa-gpio.h | 2 ++ sound/aoa/core/gpio-feature.c | 17 ++++++++++++++++- sound/aoa/fabrics/layout.c | 7 +++++++ 3 files changed, 25 insertions(+), 1 deletion(-) --- wireless-testing.orig/sound/aoa/aoa-gpio.h 2009-01-15 18:09:48.000000000 +0100 +++ wireless-testing/sound/aoa/aoa-gpio.h 2009-01-15 18:15:47.000000000 +0100 @@ -34,10 +34,12 @@ struct gpio_methods { void (*set_headphone)(struct gpio_runtime *rt, int on); void (*set_speakers)(struct gpio_runtime *rt, int on); void (*set_lineout)(struct gpio_runtime *rt, int on); + void (*set_master)(struct gpio_runtime *rt, int on); int (*get_headphone)(struct gpio_runtime *rt); int (*get_speakers)(struct gpio_runtime *rt); int (*get_lineout)(struct gpio_runtime *rt); + int (*get_master)(struct gpio_runtime *rt); void (*set_hw_reset)(struct gpio_runtime *rt, int on); --- wireless-testing.orig/sound/aoa/core/gpio-feature.c 2009-01-15 18:09:48.000000000 +0100 +++ wireless-testing/sound/aoa/core/gpio-feature.c 2009-01-15 18:18:09.000000000 +0100 @@ -14,7 +14,7 @@ #include <linux/interrupt.h> #include "../aoa.h" -/* TODO: these are 20 global variables +/* TODO: these are lots of global variables * that aren't used on most machines... * Move them into a dynamically allocated * structure and use that. @@ -23,6 +23,7 @@ /* these are the GPIO numbers (register addresses as offsets into * the GPIO space) */ static int headphone_mute_gpio; +static int master_mute_gpio; static int amp_mute_gpio; static int lineout_mute_gpio; static int hw_reset_gpio; @@ -32,6 +33,7 @@ static int linein_detect_gpio; /* see the SWITCH_GPIO macro */ static int headphone_mute_gpio_activestate; +static int master_mute_gpio_activestate; static int amp_mute_gpio_activestate; static int lineout_mute_gpio_activestate; static int hw_reset_gpio_activestate; @@ -156,6 +158,7 @@ static int ftr_gpio_get_##name(struct gp FTR_GPIO(headphone, 0); FTR_GPIO(amp, 1); FTR_GPIO(lineout, 2); +FTR_GPIO(master, 3); static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) { @@ -172,6 +175,8 @@ static void ftr_gpio_set_hw_reset(struct hw_reset_gpio, v); } +static struct gpio_methods methods; + static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) { int saved; @@ -181,6 +186,8 @@ static void ftr_gpio_all_amps_off(struct ftr_gpio_set_headphone(rt, 0); ftr_gpio_set_amp(rt, 0); ftr_gpio_set_lineout(rt, 0); + if (methods.set_master) + ftr_gpio_set_master(rt, 0); rt->implementation_private = saved; } @@ -193,6 +200,8 @@ static void ftr_gpio_all_amps_restore(st ftr_gpio_set_headphone(rt, (s>>0)&1); ftr_gpio_set_amp(rt, (s>>1)&1); ftr_gpio_set_lineout(rt, (s>>2)&1); + if (methods.set_master) + ftr_gpio_set_master(rt, (s>>3)&1); } static void ftr_handle_notify(struct work_struct *work) @@ -231,6 +240,12 @@ static void ftr_gpio_init(struct gpio_ru get_gpio("hw-reset", "audio-hw-reset", &hw_reset_gpio, &hw_reset_gpio_activestate); + if (get_gpio("master-mute", NULL, + &master_mute_gpio, + &master_mute_gpio_activestate)) { + methods.set_master = ftr_gpio_set_master; + methods.get_master = ftr_gpio_get_master; + } headphone_detect_node = get_gpio("headphone-detect", NULL, &headphone_detect_gpio, --- wireless-testing.orig/sound/aoa/fabrics/layout.c 2009-01-15 18:15:28.000000000 +0100 +++ wireless-testing/sound/aoa/fabrics/layout.c 2009-01-15 18:15:47.000000000 +0100 @@ -600,6 +600,7 @@ struct layout_dev { struct snd_kcontrol *headphone_ctrl; struct snd_kcontrol *lineout_ctrl; struct snd_kcontrol *speaker_ctrl; + struct snd_kcontrol *master_ctrl; struct snd_kcontrol *headphone_detected_ctrl; struct snd_kcontrol *lineout_detected_ctrl; @@ -651,6 +652,7 @@ static struct snd_kcontrol_new n##_ctl = AMP_CONTROL(headphone, "Headphone Switch"); AMP_CONTROL(speakers, "Speakers Switch"); AMP_CONTROL(lineout, "Line-Out Switch"); +AMP_CONTROL(master, "Master Switch"); static int detect_choice_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -891,6 +893,11 @@ static void layout_attached_codec(struct lineout = codec->gpio->methods->get_detect(codec->gpio, AOA_NOTIFY_LINE_OUT); + if (codec->gpio->methods->set_master) { + ctl = snd_ctl_new1(&master_ctl, codec->gpio); + ldev->master_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } while (cc->connected) { if (cc->connected & CC_SPEAKERS) { if (headphones <= 0 && lineout <= 0)

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[alsa-devel] Mac Pro problem with USB soundcard
by James Ward 15 Jan '09

15 Jan '09

I'm running 64 Bit Sabayon Linux 4.0 r1 on a Mac Pro. I have connected a Creative USB sound card and would like to use it as the default. I tried these two .asoundrc files suggested by gnubien on #alsa: pcm.!default { type hw card 1 } ctl.!default { type hw card 1 } and: created by wishie on #alsa 10-13-07 #Set the card you want to use here.. this example is for hw:1,0 pcm.snd_card { type hw card 1 #change "card 1" to the card number listed for your usb soundcard using cat /proc/asound/cards output device 0 } pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 ipc_key_add_uid false ipc_perm 0660 slave { pcm "snd_card" rate 48000 period_time 0 period_size 1024 buffer_time 0 buffer_size 5120 } } Either of these .asoundrc files result in the USB soundcard going away. I have to unplug it and plug it back in to get it back. Please help. James -- Ward... James Ward Linux System Administrator System Administration Phone: 520-290-0910 ext 268 Fax: 520-546-3442 ICQ: 201663408 Jabber: james(a)jabber.chinoc.net

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[alsa-devel] RFC: snd_card_create() function
by Takashi Iwai 15 Jan '09

15 Jan '09

Hi, so far, we use snd_card_new() function to create a card instance. A known problem regarding this API is that it doesn't return a proper error code, thus the probe callback always returns -ENOMEM (or whatever the driver defines) no matter which error occurred actually. A typical case is the card slot conflict. Even in such a case, it returns -ENOMEM, which may mislead the user. For fixing this situation, I'd like to convert from snd_card_new() to a new function, snd_card_create(): int snd_card_create(int idx, const char *id, struct module *module, int extra_size, struct snd_card **card_ret); and provides snd_card_new() as a wrapper with deprecated flag: static inline __deprecated struct snd_card *snd_card_new(int idx, const char *id, struct module *module, int extra_size) { struct snd_card *card; if (snd_card_create(idx, id, module, extra_size, &card) < 0) return NULL; return card; } The merit of creating a new function is that the older and out-of-tree codes (like xfi) work without modification. There is alternative option, namely to use PTR_ERR() and keep the function as is. But this has no advantage in practice -- it can't detect the old API at build time, and it may Oops. The preliminary patches are found in test/snd_card_new-err branch of sound git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git test/snd_card_new-err If anyone finds problems with this, please let me know. thanks, Takashi

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[alsa-devel] usb-audio extension unit
by James Courtier-Dutton 15 Jan '09

15 Jan '09

Hi, Is there any chance that someone could add get/set functions for usb audio "extension units". This would make adding support for usb devices such as the emu usb products a lot easier to do. Kind Regards James

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[alsa-devel] [PATCH 0/1] ASoC: fix the dapm_mux_update_power function for value_mux
by Peter Ujfalusi 15 Jan '09

15 Jan '09

Hello, To be honest I don't know how I have missed this from the previous series with the cleanup for the value_mux type. the dapm_mux_update_power function needs to check for both snd_soc_dapm_mux and snd_soc_dapm_value_mux. --- Peter Ujfalusi (1): ASoC: Fix the power update function for snd_soc_dapm_value_mux sound/soc/soc-dapm.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-)

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Re: [alsa-devel] [Fwd: Re: Fwd: audio ovolume problem on powerbook G3 Lombard (PPC740)]
by Risto Suominen 15 Jan '09

15 Jan '09

2009/1/8 Takashi Iwai <tiwai(a)suse.de>: > At Thu, 8 Jan 2009 14:06:05 +0200, > Risto Suominen wrote: >> >> 2009/1/8, Takashi Iwai <tiwai(a)suse.de>: >> > >> > We may use a virtual master control when all slave controls have the >> > identical volume resolution. >> > >> >> Sounds good. Is it something configurable, so there is no need to >> change the code? > > Look at sound/core/vmaster.c. > > > Takashi > Ok, I've tried to implement this on snd-powermac. Takashi, looks like there is a side effect: plugging in headphone doesn't show on alsamixer, even though it works, i.e. the speaker is silenced. Or have I forgotten something in my implementation? http://ristosu.wippiespace.com/pub/alsa-awacs-1.0.18a-p13.diff Richard, maybe you can try this? You should adjust Speaker and Line out to 100% in order to have full scale in Master. Risto

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