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[alsa-devel] 2.6.26.[6|7] vs. rt11 vs. alsa (usb) midi
by Fernando Lopez-Lezcano 07 Nov '08

07 Nov '08

Hi... anyone out there also having problem with this combination? 2.6.26.7 or 2.6.26.6 with 2.6.26.6-rt11 and alsa MIDI It looks to me like alsa midi is broken. Probably an interaction with the realtime kernel patch. A Fedora kernel based on 2.6.26.6 (2.6.26.6-49 on fc8) works fine, a very similarly patched kernel that includes the realtime patch does not. The configurations for both are almost the same (today I tried to change some of the most obvious differences and rebuild, with the same result). No problems with 2.6.24.7-rt21. Test: boot, login, plugin an external usb keyboard (Yamaha P250), it is recognized, use qjackctl alsa patchbay to connect it to kmidimon, kmidimon sees only _one_ midi message and that's it. Kmidimon can't be killed after that. -- Fernando

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Re: [alsa-devel] supported sound cards
by Edward Terry 06 Nov '08

06 Nov '08

Clemens Ladisch wrote: > The only chip where hardware mixing is supported is Creative's Emu10k1 > (snd-emu10k1 driver), used on the SB Live!, most Audigy and certain > low-end X-Fi cards. (Cards with the 'real' X-Fi chip do _not_ work > well in Linux.) > > Other supported chips are ICE1724 (M-Audio Delta 1010(LT), DiO 2496, > 66, 44, 410, Audiophile 24/96; Digigram VX442; TerraTec EWX 24/96, > EWS 88MT/D, DMX 6Fire, Phase 88; Hoontech SoundTrack DSP 24/Value/ > Media7.1; Event EZ8; Lionstracs Mediastation, Terrasoniq TS 88) and > VT1720/24 (AMP AUDIO2000; M-Audio Revolution 5.1, 7.1, Audiophile 192; > TerraTec Aureon 5.1 Sky, 7.1 Space/Universe, Phase 22/28; > Onkyo SE-90PCI, SE-200PCI; AudioTrak Prodigy 192, 7.1 (HIFI/LT/XT), > HD2; Hercules Fortissimo IV; ESI Juli@; Pontis MS300; EGO-SYS > WaveTerminal 192M). Thanks. I'll look over these cards. > The most high-end supported cards are probably the Asus Xonar cards. > The D2/D2X have even slightly better audio quality than the best X-Fi, > and, as the obviously most important feature, colorfully illuminated > jacks. The various Dolby features are done in software in the Windows > driver and are not supported in Linux. If you need a PCI-E card, the > Xonar DX or D2X are your only choice. What about the Asus Xonar HDAV1.3? John Rigg wrote: >> The most high-end supported cards are probably the Asus Xonar cards. > > `High end' means different things in different situations of course. > RME HDSP series are probably the `highest end' cards supported, but > they need external converters, so they're probably unsuitable here. The Xonar cards do look interesting, but I'd also like to carry one higher-end card. I'm looking for a single-slot PCI Express solution that works as a standalone sound card, i.e. without any mandatory external devices (though optional ones are fine). Thanks for the information. Edward

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Re: [alsa-devel] supported sound cards
by Edward Terry 06 Nov '08

06 Nov '08

Clemens Ladisch wrote: > The only chip where hardware mixing is supported is Creative's Emu10k1 > (snd-emu10k1 driver), used on the SB Live!, most Audigy and certain > low-end X-Fi cards. (Cards with the 'real' X-Fi chip do _not_ work > well in Linux.) > > Other supported chips are ICE1724 (M-Audio Delta 1010(LT), DiO 2496, > 66, 44, 410, Audiophile 24/96; Digigram VX442; TerraTec EWX 24/96, > EWS 88MT/D, DMX 6Fire, Phase 88; Hoontech SoundTrack DSP 24/Value/ > Media7.1; Event EZ8; Lionstracs Mediastation, Terrasoniq TS 88) and > VT1720/24 (AMP AUDIO2000; M-Audio Revolution 5.1, 7.1, Audiophile 192; > TerraTec Aureon 5.1 Sky, 7.1 Space/Universe, Phase 22/28; > Onkyo SE-90PCI, SE-200PCI; AudioTrak Prodigy 192, 7.1 (HIFI/LT/XT), > HD2; Hercules Fortissimo IV; ESI Juli@; Pontis MS300; EGO-SYS > WaveTerminal 192M). Thanks. I'll look over these cards. > The most high-end supported cards are probably the Asus Xonar cards. > The D2/D2X have even slightly better audio quality than the best X-Fi, > and, as the obviously most important feature, colorfully illuminated > jacks. The various Dolby features are done in software in the Windows > driver and are not supported in Linux. If you need a PCI-E card, the > Xonar DX or D2X are your only choice. What about the Asus Xonar HDAV1.3? John Rigg wrote: >> The most high-end supported cards are probably the Asus Xonar cards. > > `High end' means different things in different situations of course. > RME HDSP series are probably the `highest end' cards supported, but > they need external converters, so they're probably unsuitable here. The Xonar cards do look interesting, but I'd also like to carry one higher-end card. I'm looking for a single-slot PCI Express solution that works as a standalone sound card, i.e. without any mandatory external devices (though optional ones are fine). Thanks for the information. Edward

1 0

[alsa-devel] [PATCH] Add lifebook model for Realtek ALC269
by Tony Vroon 06 Nov '08

06 Nov '08

The widget layout of the Fujitsu Lifebook S6420 (which is ICH9M-based and uses an ALC269) is similar but not identical to the Lifebook S6410/E8410 (which are ICH8M-based and use an ALC262). It is named lifebook as fujitsu is in use for Amilo machines. This builds on the Quanta FL1 work and supports all analog inputs & outputs that I am aware of. Microphone autoswitch is implemented. The laptop mic port takes precedence over the dock mic port if both happen to have a jack plugged in. This made sense to me as a design decision (imagine a presentation environment with the dock fully wired in and the presenter quickly wanting to override the mic with a headset). There is mention of a digital audio path on the codec graph, so perhaps the headphone socket is dual-function analog/digital. I will follow up with another patch if I can acquire equipment to test this. Signed-off-by: Tony Vroon <tony(a)linx.net> --- patch_realtek.c.orig 2008-11-06 14:05:06.000000000 +0000 +++ patch_realtek.c 2008-11-06 14:19:51.000000000 +0000 @@ -132,6 +132,7 @@ ALC269_ASUS_EEEPC_P703, ALC269_ASUS_EEEPC_P901, ALC269_FUJITSU, + ALC269_LIFEBOOK, ALC269_AUTO, ALC269_MODEL_LAST /* last tag */ }; @@ -11703,6 +11704,31 @@ { } }; +static struct snd_kcontrol_new alc269_lifebook_mixer[] = { + /* output mixer control */ + HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc268_acer_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), + { } +}; + /* bind volumes of both NID 0x0c and 0x0d */ static struct hda_bind_ctls alc269_epc_bind_vol = { .ops = &snd_hda_bind_vol, @@ -11753,6 +11779,20 @@ { } }; +static struct hda_verb alc269_lifebook_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + { } +}; + /* toggle speaker-output according to the hp-jack state */ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) { @@ -11778,6 +11818,37 @@ AC_VERB_SET_PROC_COEF, 0x480); } +/* toggle speaker-output according to the hp-jacks state */ +static void alc269_lifebook_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + /* Check laptop headphone socket */ + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Check port replicator headphone socket */ + present |= snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + bits = present ? AMP_IN_MUTE(0) : 0; + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); + + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x680); + + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x480); +} + static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec) { unsigned int present; @@ -11788,6 +11859,29 @@ AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1); } +static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec) +{ + unsigned int present_laptop; + unsigned int present_dock; + + present_laptop = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + present_dock = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Laptop mic port overrides dock mic port, design decision */ + if (present_dock) + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, 0x3); + if (present_laptop) + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, 0x0); + if (!present_dock && !present_laptop) + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, 0x1); +} + static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -11797,12 +11891,27 @@ alc269_quanta_fl1_mic_automute(codec); } +static void alc269_lifebook_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc269_lifebook_speaker_automute(codec); + if ((res >> 26) == ALC880_MIC_EVENT) + alc269_lifebook_mic_autoswitch(codec); +} + static void alc269_quanta_fl1_init_hook(struct hda_codec *codec) { alc269_quanta_fl1_speaker_automute(codec); alc269_quanta_fl1_mic_automute(codec); } +static void alc269_lifebook_init_hook(struct hda_codec *codec) +{ + alc269_lifebook_speaker_automute(codec); + alc269_lifebook_mic_autoswitch(codec); +} + static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x23, AC_VERB_SET_CONNECT_SEL, 0x05}, @@ -12156,7 +12265,8 @@ [ALC269_QUANTA_FL1] = "quanta", [ALC269_ASUS_EEEPC_P703] = "eeepc-p703", [ALC269_ASUS_EEEPC_P901] = "eeepc-p901", - [ALC269_FUJITSU] = "fujitsu" + [ALC269_FUJITSU] = "fujitsu", + [ALC269_LIFEBOOK] = "lifebook" }; static struct snd_pci_quirk alc269_cfg_tbl[] = { @@ -12168,6 +12278,7 @@ SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101", ALC269_ASUS_EEEPC_P901), SND_PCI_QUIRK(0x1734, 0x115d, "FSC Amilo", ALC269_FUJITSU), + SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook ICH9M-based", ALC269_LIFEBOOK), {} }; @@ -12236,6 +12347,18 @@ .unsol_event = alc269_eeepc_dmic_unsol_event, .init_hook = alc269_eeepc_dmic_inithook, }, + [ALC269_LIFEBOOK] = { + .mixers = { alc269_lifebook_mixer }, + .init_verbs = { alc269_init_verbs, alc269_lifebook_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_capture_source, + .unsol_event = alc269_lifebook_unsol_event, + .init_hook = alc269_lifebook_init_hook, + }, }; static int patch_alc269(struct hda_codec *codec)

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[alsa-devel] supported sound cards
by Edward Terry 06 Nov '08

06 Nov '08

I'm starting a business selling Linux-based computers, and I'm looking for a high-end sound card that works well under Linux. I don't have a specific use in mind; this would be for a wide range of end-users, from audiophiles to gamers. Is there a particular high-end sound card that stands out as working well under Linux? If not, what cards should I consider? I appreciate any information or suggestions. Edward

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[alsa-devel] [PATCH] Add fujitsu model for Realtek ALC269
by Tony Vroon 06 Nov '08

06 Nov '08

The widget layout of the Fujitsu Lifebook S6420 (which is ICH9M-based and uses an ALC269) is similar but not identical to the Lifebook S6410/E8410 (which are ICH8M-based and use an ALC262). This is builds on the Quanta FL1 work and supports all analog inputs & outputs that I am aware of. The microphone automute is more of an auto-switch as there are three inputs. The laptop mic port takes precedence over the dock mic port if both happen to have a jack plugged in. This made sense to me as a design decision (imagine a presentation environment with the dock fully wired in and the presenter quickly wanting to override the mic with a headset). There is mention of a digital audio path on the codec graph, so perhaps the headphone socket is dual-function analog/digital. I will follow up with another patch if I can acquire equipment to test this. This applies on top of current linus git, let me know if this is inconvenient and you'd like it based on something else. Signed-off-by: Tony Vroon <tony(a)linx.net> --- linux-2.6/sound/pci/hda/patch_realtek.c.orig 2008-11-05 18:34:26.000000000 +0000 +++ linux-2.6/sound/pci/hda/patch_realtek.c 2008-11-05 20:20:47.000000000 +0000 @@ -130,6 +130,7 @@ ALC269_QUANTA_FL1, ALC269_ASUS_EEEPC_P703, ALC269_ASUS_EEEPC_P901, + ALC269_FUJITSU, ALC269_AUTO, ALC269_MODEL_LAST /* last tag */ }; @@ -11892,6 +11893,31 @@ { } }; +static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { + /* output mixer control */ + HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc268_acer_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), + { } +}; + /* bind volumes of both NID 0x0c and 0x0d */ static struct hda_bind_ctls alc269_epc_bind_vol = { .ops = &snd_hda_bind_vol, @@ -11952,6 +11978,20 @@ { } }; +static struct hda_verb alc269_fujitsu_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + { } +}; + /* toggle speaker-output according to the hp-jack state */ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) { @@ -11977,6 +12017,37 @@ AC_VERB_SET_PROC_COEF, 0x480); } +/* toggle speaker-output according to the hp-jack state */ +static void alc269_fujitsu_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + /* Check laptop headphone socket */ + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Check port replicator headphone socket */ + present |= snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + bits = present ? AMP_IN_MUTE(0) : 0; + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); + + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x680); + + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x480); +} + static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec) { unsigned int present; @@ -11987,6 +12058,29 @@ AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1); } +static void alc269_fujitsu_mic_automute(struct hda_codec *codec) +{ + unsigned int present_laptop; + unsigned int present_dock; + + present_laptop = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + present_dock = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + + /* Laptop mic port overrides dock mic port, design decision */ + if (present_dock) + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, 0x3); + if (present_laptop) + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, 0x0); + if (!present_dock && !present_laptop) + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, 0x1); +} + static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -11996,12 +12090,27 @@ alc269_quanta_fl1_mic_automute(codec); } +static void alc269_fujitsu_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc269_fujitsu_speaker_automute(codec); + if ((res >> 26) == ALC880_MIC_EVENT) + alc269_fujitsu_mic_automute(codec); +} + static void alc269_quanta_fl1_init_hook(struct hda_codec *codec) { alc269_quanta_fl1_speaker_automute(codec); alc269_quanta_fl1_mic_automute(codec); } +static void alc269_fujitsu_init_hook(struct hda_codec *codec) +{ + alc269_fujitsu_speaker_automute(codec); + alc269_fujitsu_mic_automute(codec); +} + static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x23, AC_VERB_SET_CONNECT_SEL, 0x05}, @@ -12354,7 +12463,8 @@ [ALC269_BASIC] = "basic", [ALC269_QUANTA_FL1] = "quanta", [ALC269_ASUS_EEEPC_P703] = "eeepc-p703", - [ALC269_ASUS_EEEPC_P901] = "eeepc-p901" + [ALC269_ASUS_EEEPC_P901] = "eeepc-p901", + [ALC269_FUJITSU] = "fujitsu" }; static struct snd_pci_quirk alc269_cfg_tbl[] = { @@ -12365,6 +12475,7 @@ ALC269_ASUS_EEEPC_P901), SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101", ALC269_ASUS_EEEPC_P901), + SND_PCI_QUIRK(0x10cf, 0x1475, "Fujitsu ICH9M-based", ALC269_FUJITSU), {} }; @@ -12417,6 +12528,18 @@ .unsol_event = alc269_eeepc_dmic_unsol_event, .init_hook = alc269_eeepc_dmic_inithook, }, + [ALC269_FUJITSU] = { + .mixers = { alc269_fujitsu_mixer }, + .init_verbs = { alc269_init_verbs, alc269_fujitsu_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_capture_source, + .unsol_event = alc269_fujitsu_unsol_event, + .init_hook = alc269_fujitsu_init_hook, + }, }; static int patch_alc269(struct hda_codec *codec)

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[alsa-devel] Any issues with GETOSPACE OSS ioctl?
by Timur Tabi 06 Nov '08

06 Nov '08

I have a customer who reports that when his OSS application makes several write() followed by SNDCTL_DSP_GETOSPACE, the returned value is not decreasing, which implies that the write() commands are not going through. I've looked at the OSS documentation for this ioctl (http://manuals.opensound.com/developer/SNDCTL_DSP_GETOSPACE.html), and it implies that it is not reliable. But since the customer is using ALSA (i.e. ALSA's OSS emulation), I don't know if that applies. So I'd like to know how the 'bytes' return value is calculate in ALSA. -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 06 Nov '08

06 Nov '08

The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc: Takashi Iwai (1): ALSA: ASoC - Remove unnecessary inclusion of linux/version.h are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai This series drops the TLV320AIC23 patch for now and makes the Palm driver non-modular for the time being due to the platform data dodge being used. David Anders (1): ASoC: Add new parameter to s3c24xx_pcm_enqueue Grazvydas Ignotas (1): ALSA: ASoC: TWL4030 codec - fix 256*Fs clock Marek Vasut (1): ASoC: Add Palm/PXA27x unified ASoC audio driver Mark Brown (1): ASoC: Remove core version number arch/arm/mach-pxa/include/mach/palmasoc.h | 13 ++ include/sound/soc.h | 2 - sound/soc/codecs/twl4030.c | 4 +- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/palm27x.c | 269 +++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c24xx-pcm.c | 12 +- sound/soc/soc-core.c | 1 - 8 files changed, 303 insertions(+), 9 deletions(-) create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h create mode 100644 sound/soc/pxa/palm27x.c

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[alsa-devel] [PATCH] ALSA: ASoC: TWL4030 codec - fix 256*Fs clock
by Grazvydas Ignotas 06 Nov '08

06 Nov '08

According to TRM, 256*Fs clock output should be enabled when TWL4030 is in slave mode, not master. This allows sound to work on OMAP3 Pandora, which uses 256*Fs clock. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/codecs/twl4030.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ee2f0d3..90f3b4d 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -469,11 +469,11 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: format &= ~(TWL4030_AIF_SLAVE_EN); - format |= TWL4030_CLK256FS_EN; + format &= ~(TWL4030_CLK256FS_EN); break; case SND_SOC_DAIFMT_CBS_CFS: - format &= ~(TWL4030_CLK256FS_EN); format |= TWL4030_AIF_SLAVE_EN; + format |= TWL4030_CLK256FS_EN; break; default: return -EINVAL; -- 1.5.4.3

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[alsa-devel] [PATCH] ASoC: Add new parameter to s3c24xx_pcm_enqueue
by David Anders 05 Nov '08

05 Nov '08

The S3C24xx dma does not allow more than one buffer to be enqueue prior to the dma transfers starting. This patch adds an additional parameter to s3c24xx_pcm_enqueue() to allow for passing an initial dma maximum load value. Signed-off-by: David Anders <danders at amltd.com> diff -urN linux-2.6.27-clean/sound/soc/s3c24xx/s3c24xx-pcm.c linux-2.6.27/sound/soc/s3c24xx/s3c24xx-pcm.c --- linux-2.6.27-clean/sound/soc/s3c24xx/s3c24xx-pcm.c 2008-11-05 08:27:23.000000000 -0600 +++ linux-2.6.27/sound/soc/s3c24xx/s3c24xx-pcm.c 2008-11-05 09:14:27.000000000 -0600 @@ -78,7 +78,8 @@ * place a dma buffer onto the queue for the dma system * to handle. */ -static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream) +static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream, + int dma_max) { struct s3c24xx_runtime_data *prtd = substream->runtime->private_data; dma_addr_t pos = prtd->dma_pos; @@ -86,7 +87,10 @@ DBG("Entered %s\n", __func__); - while (prtd->dma_loaded < prtd->dma_limit) { + if (!dma_max) + dma_max = prtd->dma_limit; + + while (prtd->dma_loaded < dma_max) { unsigned long len = prtd->dma_period; DBG("dma_loaded: %d\n", prtd->dma_loaded); @@ -132,7 +136,7 @@ spin_lock(&prtd->lock); if (prtd->state & ST_RUNNING) { prtd->dma_loaded--; - s3c24xx_pcm_enqueue(substream); + s3c24xx_pcm_enqueue(substream, 0); } spin_unlock(&prtd->lock); @@ -249,7 +253,7 @@ prtd->dma_pos = prtd->dma_start; /* enqueue dma buffers */ - s3c24xx_pcm_enqueue(substream); + s3c24xx_pcm_enqueue(substream, 1); return ret; }

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