- Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] alsa-lib: Make sure SND_PCM_NO_xxx flags don't get lost when nonblocking mode is enabled
by Lennart Poettering 31 Aug '08

31 Aug '08

The plug PCM copies the 'mode' field from the slave PCM. If blocking mode is enabled for the plug PCM the mode is subsequently overwritten with the original requested 'mode'. If non-blocking mode is requested this does not happen. Because the hw PCM synthesizes the 'mode' from the actual file descriptor flags no SND_PCM_NO_xxx will ever be set for it. This has the effect that the 'mode' of the plug PCM will also not include those flags anymore -- unless they are overwritten as mentioned above. This basically means SND_PCM_NO_xxx is ignored for plug:hw:4711 style device strings opened in non-blocking mode. You can easily test this with "aplay --channels 7 --disable-channels -f S16_LE -r 44100 -D plug:hw:0" on a device that cannot do 7 channels. Normally this call should fail, however if you add "-N" to the command line this call will succeed. This patch simply copies the SND_PCM_NO_xxx flags back into the 'mode' field in case we don't overwrite it with the original anyway. Probably closes bug 3571 for good. Lennart --- src/pcm/pcm_hw.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 1e1889c..c8d9a62 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1507,7 +1507,13 @@ int _snd_pcm_hw_open(snd_pcm_t **pcmp, const char *name, /* revert to blocking mode for read/write access */ snd_pcm_hw_nonblock(*pcmp, 0); (*pcmp)->mode = mode; - } + } else + /* make sure the SND_PCM_NO_xxx flags don't get lost on the + * way */ + (*pcmp)->mode |= mode & (SND_PCM_NO_AUTO_RESAMPLE| + SND_PCM_NO_AUTO_CHANNELS| + SND_PCM_NO_AUTO_FORMAT| + SND_PCM_NO_SOFTVOL); hw = (*pcmp)->private_data; if (format != SND_PCM_FORMAT_UNKNOWN) -- 1.6.0.1 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

1 0

Re: [alsa-devel] ALSA, select() question.
by William Estrada 29 Aug '08

29 Aug '08

Jaroslav, I got you Email address from http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg09605.html. <http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg09605.html> I am trying to write a C program that simply reads from a ALSA device and then writes back to it. The program work with one major problem, there is a 4.1 second delay between the read and the write. This delay is caused by the select() not returning at the right time. In you post you said to use: > Yes, but you have to create revents for snd_seq_poll_descriptors_revents() > to get the right event then. Don't use values from select() directly. > > Jaroslav > Do you have some sample code that I could look at so that I could fix my problem? -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

[alsa-devel] Issues with alsa-info.sh - Trying to track down the problem.
by Travis Place 29 Aug '08

29 Aug '08

Since the move to use alsa-project.org as the main place for uploads, some users have reported that no URL is being printed at the end of the script. I am trying to track down the problem, but would appreciate some help if possible. Thanks in advance, Travis Place (wishie)

2 4

[alsa-devel] [PATCH] ice1724 MIDI on PHASE 22 and PHASE 28 and a few minor fixes
by Vedran Miletić 29 Aug '08

29 Aug '08

Changelog: * added info about PHASE 28 (as far as I could collect, I don't have a card) * added info about TS22PCI and new revisions of PHASE 22 * disable 192k on PHASE 22 as AK4524 I2S doesn't support it * enable midi for both PHASE 22 and PHASE 28 * enable SPDIF reciever on PHASE 22 MIDI works for me, I have tried loopback and it plays good. Thanks for help Pavel and Takashi. diff --git a/pci/ice1712/phase.c b/pci/ice1712/phase.c index f5acdee..6e596e9 100644 --- a/pci/ice1712/phase.c +++ b/pci/ice1712/phase.c @@ -24,7 +24,8 @@ /* PHASE 22 overview: * Audio controller: VIA Envy24HT-S (slightly trimmed down version of Envy24HT) * Analog chip: AK4524 (partially via Philip's 74HCT125) - * Digital receiver: CS8414-CS (not supported in this release) + * Digital receiver: CS8414-CS (supported in this release) + * PHASE 22 revision 2.0 and Terrasoniq/Musonik TS22PCI have CS8416 (support status unknown, please test and report) * * Envy connects to AK4524 * - CS directly from GPIO 10 @@ -33,6 +34,12 @@ * CDTI may be completely blocked by 74HCT125's gate #1 controlled by GPIO 3 */ +/* PHASE 28 overview: + * Audio controller: VIA Envy24HT (full untrimmed version, 8 ins 8 outs) + * Analog chip: WM8770 (8 channel DAC, 2 channel ADC) + * Digital receiver: CS8414-CS (supported in this release) + */ + #include <asm/io.h> #include <linux/delay.h> #include <linux/interrupt.h> @@ -161,9 +168,9 @@ static int __devinit phase22_add_controls(struct snd_ice1712 *ice) } static unsigned char phase22_eeprom[] __devinitdata = { - [ICE_EEP2_SYSCONF] = 0x00, /* 1xADC, 1xDACs */ + [ICE_EEP2_SYSCONF] = 0x28, /* clock 512, mpu 401, spdif-in/1xADC, 1xDACs */ [ICE_EEP2_ACLINK] = 0x80, /* I2S */ - [ICE_EEP2_I2S] = 0xf8, /* vol, 96k, 24bit */ + [ICE_EEP2_I2S] = 0xf0, /* vol, 96k, 24bit */ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */ [ICE_EEP2_GPIO_DIR] = 0xff, [ICE_EEP2_GPIO_DIR1] = 0xff, @@ -177,7 +184,7 @@ static unsigned char phase22_eeprom[] __devinitdata = { }; static unsigned char phase28_eeprom[] __devinitdata = { - [ICE_EEP2_SYSCONF] = 0x0b, /* clock 512, spdif-in/ADC, 4DACs */ + [ICE_EEP2_SYSCONF] = 0x2b, /* clock 512, mpu401, spdif-in/1xADC, 4xDACs */ [ICE_EEP2_ACLINK] = 0x80, /* I2S */ [ICE_EEP2_I2S] = 0xfc, /* vol, 96k, 24bit, 192k */ [ICE_EEP2_SPDIF] = 0xc3, /* out-en, out-int, spdif-in */ -- Vedran Miletić

2 9

[alsa-devel] [PATCH] ice1724: enable MIDI on Terratec Aureon 7.1 Universe
by Maximilian Rehkopf 29 Aug '08

29 Aug '08

This enables MIDI on Terratec Aureon 7.1 Universe cards specifically. Apparently the other envy24ht-based Aureon cards do not have MIDI ports, hence I added a Universe specific eeprom array. The newer cards (Aureon 7.1 PCI / 5.1 PCI/Fun) have MIDI but use a CMI DSP so this driver does not concern them at all. I am new to git, do I have to perform any git operations regarding the patch myself? Signed-off-by: Maximilian Rehkopf <otakon(a)gmx.net> -- Maximilian Rehkopf <otakon(a)gmx.net>

3 3

[alsa-devel] PATCH - via82xx vt1618 7.1 Audio Support
by John L. Utz III 29 Aug '08

29 Aug '08

1. Implement 7.1 Output for vt1618 codec. 2. Fix typos in comments from my previous 1617a patch. Signed-off-by: John L. Utz III <john.utz(a)dmx.com> diff --git a/pci/ac97/ac97_codec.c b/pci/ac97/ac97_codec.c index d0023e9..89f3050 100644 --- a/pci/ac97/ac97_codec.c +++ b/pci/ac97/ac97_codec.c @@ -168,7 +168,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -{ 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, +{ 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, @@ -609,7 +609,6 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0); - static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; static const char* std_3d_path[] = {"pre 3D", "post 3D"}; static const char* std_mix[] = {"Mix", "Mic"}; diff --git a/pci/ac97/ac97_patch.c b/pci/ac97/ac97_patch.c index bb028f8..7d5abf5 100644 --- a/pci/ac97/ac97_patch.c +++ b/pci/ac97/ac97_patch.c @@ -3465,7 +3465,7 @@ static int patch_vt1616(struct snd_ac97 * ac97) /* * unfortunately, the vt1617a stashes the twiddlers required for - * nooding the i/o jacks on 2 different regs. * thameans that we cant + * noodling the i/o jacks on 2 different regs. that means that we cant * use the easy way provided by AC97_ENUM_DOUBLE() we have to write * are own funcs. * @@ -3498,7 +3498,7 @@ static int snd_ac97_vt1617a_smart51_get(struct snd_kcontrol *kcontrol, pac97 = snd_kcontrol_chip(kcontrol); /* grab codec handle */ - /* grab our desirec bits, then mash them together in a manner + /* grab our desired bits, then mash them together in a manner * consistent with Table 6 on page 17 in the 1617a docs */ usSM51 = snd_ac97_read(pac97, 0x7a) >> 14; @@ -3576,6 +3576,201 @@ int patch_vt1617a(struct snd_ac97 * ac97) return err; } +/* VIA VT1618 8 CHANNEL AC97 CODEC + * + * VIA implements 'Smart 5.1' completely differently on the 1618 than + * it does on the 1617a. awesome! They seem to have sourced this + * particular revision of the technology from somebody else, it's + * called Universal Audio Jack and it shows up on some other folk's chips + * as well. + * + * ordering in this list reflects vt1618 docs for Reg 60h and + * the block diagram, DACs are as follows: + * + * OUT_O -> Front, + * OUT_1 -> Surround, + * OUT_2 -> C/LFE + * + * Unlike the 1617a, each OUT has a consistent set of mappings + * for all bitpatterns other than 00: + * + * 01 Unmixed Output + * 10 Line In + * 11 Mic In + * + * Special Case of 00: + * + * OUT_0 Mixed Output + * OUT_1 Reserved + * OUT_2 Reserved + * + * I have no idea what the hell Reserved does, but on an MSI + * CN700T, i have to set it to get 5.1 output - YMMV, bad + * shit may happen. + * + * If other chips use Universal Audio Jack, then this code might be applicable + * to them. + */ + +struct vt1618_uaj_item { + unsigned short mask; + unsigned short shift; + const char *items[4]; +}; + +/* This list reflects the vt1618 docs for Vendor Defined Register 0x60. */ + +static struct vt1618_uaj_item vt1618_uaj[3] = { + { + /* speaker jack */ + .mask = 0x03, + .shift = 0, + .items = { + "Speaker Out", "DAC Unmixed Out", "Line In", "Mic In" + } + }, + { + /* line jack */ + .mask = 0x0c, + .shift = 2, + .items = { + "Surround Out", "DAC Unmixed Out", "Line In", "Mic In" + } + }, + { + /* mic jack */ + .mask = 0x30, + .shift = 4, + .items = { + "Center LFE Out", "DAC Unmixed Out", "Line In", "Mic In" + }, + }, +}; + +static int snd_ac97_vt1618_UAJ_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return ac97_enum_text_info(kcontrol, uinfo, + vt1618_uaj[kcontrol->private_value].items, + 4); +} + +/* All of the vt1618 Universal Audio Jack twiddlers are on + Vendor Defined Register 0x60, page 0. The bits, and thus + the mask, are the only thing that changes */ + +static int snd_ac97_vt1618_UAJ_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned short datpag, uaj; + + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + mutex_lock(&pac97->page_mutex); + + datpag = snd_ac97_read(pac97, AC97_INT_PAGING) & AC97_PAGE_MASK; + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, 0); + + uaj = snd_ac97_read(pac97, 0x60) & + vt1618_uaj[kcontrol->private_value].mask; + + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, datpag); + mutex_unlock(&pac97->page_mutex); + + ucontrol->value.enumerated.item[0] = uaj >> + vt1618_uaj[kcontrol->private_value].shift; + + return 0; +} + +static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return ac97_update_bits_page(snd_kcontrol_chip(kcontrol), 0x60, + vt1618_uaj[kcontrol->private_value].mask, + ucontrol->value.enumerated.item[0]<< + vt1618_uaj[kcontrol->private_value].shift, + 0); +} + +/* config aux in jack - not found on 3 jack motherboards or soundcards */ + +static int snd_ac97_vt1618_aux_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *txt_aux[] = {"Aux In", "Back Surr Out"}; + + return ac97_enum_text_info(kcontrol, uinfo, txt_aux, 2); +} + +static int snd_ac97_vt1618_aux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = + (snd_ac97_read(snd_kcontrol_chip(kcontrol), 0x5c) & 0x0008)>>3; + return 0; +} + +static int snd_ac97_vt1618_aux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + /* toggle surround rear dac power */ + + snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x5c, 0x0008, + ucontrol->value.enumerated.item[0] << 3); + + /* toggle aux in surround rear out jack */ + + return snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x76, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + +static const struct snd_kcontrol_new snd_ac97_controls_vt1618[] = { + AC97_SINGLE("Exchange Center/LFE", 0x5a, 8, 1, 0), + AC97_SINGLE("DC Offset", 0x5a, 10, 1, 0), + AC97_SINGLE("Soft Mute", 0x5c, 0, 1, 1), + AC97_SINGLE("Headphone Amp", 0x5c, 5, 1, 1), + AC97_DOUBLE("Back Surr Volume", 0x5e, 8, 0, 31, 1), + AC97_SINGLE("Back Surr Switch", 0x5e, 15, 1, 1), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Speaker Jack Mode", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Jack Mode", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 1 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Jack Mode", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 2 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Jack Mode", + .info = snd_ac97_vt1618_aux_info, + .get = snd_ac97_vt1618_aux_get, + .put = snd_ac97_vt1618_aux_put, + } +}; + +int patch_vt1618(struct snd_ac97 *ac97) +{ + return patch_build_controls(ac97, snd_ac97_controls_vt1618, + ARRAY_SIZE(snd_ac97_controls_vt1618)); +} + /* */ static void it2646_update_jacks(struct snd_ac97 *ac97)

2 3

Re: [alsa-devel] [alsa-cvslog] alsa-kernel: Official ALSA project GIT repository for Linux 2.6 branch, master now at v2.6.27-rc5-122-g1328f04
by Takashi Iwai 29 Aug '08

29 Aug '08

At Fri, 29 Aug 2008 11:35:25 +0200 (CEST), noreply-git(a)alsa-project.org wrote: > > commit 1328f04260822e45a2717b65a0e2bc0c9c1255be > Author: Jaroslav Kysela <perex(a)perex.cz> > AuthorDate: Fri Aug 29 11:29:39 2008 +0200 > Commit: Jaroslav Kysela <perex(a)perex.cz> > CommitDate: Fri Aug 29 11:29:39 2008 +0200 > > ALSA: intel8x0: implement ac97_clock whitelist > > The AC97 clock detection is not accurate in some cases. This patch adds > an initial whitelist for audio devices gathered from RedHat's bugzilla: > > https://bugzilla.redhat.com/show_bug.cgi?id=441087 > > As a side effect, white-listing might speedup kernel booting (AC97 clock > measuring code is not activated). > > Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> (snip) > +struct intel8x0_clock_list { > + unsigned short subvendor; > + unsigned short subdevice; > + unsigned int rate; > +}; > + > +static struct intel8x0_clock_list intel8x0_clock_list[] __devinitdata = { > + { 0x1028, 0x00be, 44100 }, /* Analog Devices AD1885 */ > + { 0x1028, 0x0177, 48000 }, /* Analog Devices AD1980 */ > + { 0x1043, 0x80f3, 48000 }, /* Analog Devices AD1985 */ > + { 0x0000, 0x0000, 00000 } /* terminator */ > +}; > + > +static int __devinit intel8x0_in_clock_list(struct intel8x0 *chip) > +{ > + struct pci_dev *pci = chip->pci; > + struct intel8x0_clock_list *wl; > + > + for (wl = intel8x0_clock_list; wl->subvendor; wl++) { > + if (wl->subvendor == pci->subsystem_vendor && > + wl->subdevice == pci->subsystem_device) { > + printk(KERN_INFO "intel8x0: white list rate for %04x:%04x is %i\n", > + pci->subsystem_vendor, > + pci->subsystem_device, wl->rate); > + chip->ac97_bus->clock = wl->rate; > + return 1; > + } > + } > + return 0; > +} Use snd_pci_quirk_lookup() please. But, a better fix would be to fix the clock measurement. AFAIK, this seems to happen when the driver is built-in. Thus I guess it's rather a problem of time-keeping core side. Takashi

1 0

Re: [alsa-devel] [PATCH] HDA: patch-realtek: toshiba dynabook SS RX1 which use ALC262
by Takashi Iwai 28 Aug '08

28 Aug '08

At Thu, 28 Aug 2008 13:40:23 +0000, Hiroshi Miura wrote: > > Recent patch for toshiba dynabook SS RX1 against recent snapshot; Thanks. Now applied and pushed to my tree. Takashi > Hiroshi > > 2008/8/26 Takashi Iwai <tiwai(a)suse.de>: > > Hi Miura-san, > > > > At Tue, 26 Aug 2008 17:39:17 +0900, > > 三浦 SPS wrote: > >> > >> Hi Takashi, > >> > >> When I test with recent tree on the Toshiba laptop, Toshiba dynabook SS RX1, > >> some mixer items are not controlled by ALSA. > >> Some mixer items appears as an OSS mixer control. > >> Chipset is HDA intel with Realtek ALC262. > >> > >> I make this patch by using as many datas and functions in existent model. > >> > >> Toshiba hda-intel ALC262 should be controlled as same model as fujitsu. > >> There is an only difference in nid of head phone jack. > >> > >> A related bugtrack id is #3386 and I also post it there. > >> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3386 > > > > Thanks. The changes look OK, but your MUA broke the embedded patch. > > > > BTW, I merged other patches to add more support in patch_realtek.c, > > and one of them is the support for Toshiba S06. > > Could you check whether it's compatible with yours? > > It seems S06 uses the digital mic instead of analog mic, and this > > can be a difference. > > > > Signed-off-by: Hiroshi Miura <miurahr(a)acm.org> > --- > --- alsa-driver-unstable/alsa-kernel/pci/hda/patch_realtek.c.orig 2008-08-27 > 06:05:19.000000000 +0000 > +++ alsa-driver-unstable/alsa-kernel/pci/hda/patch_realtek.c 2008-08-27 > 11:28:04.000000000 +0000 > @@ -103,6 +103,7 @@ enum { > ALC262_LENOVO_3000, > ALC262_NEC, > ALC262_TOSHIBA_S06, > + ALC262_TOSHIBA_RX1, > ALC262_AUTO, > ALC262_MODEL_LAST /* last tag */ > }; > @@ -9705,6 +9706,25 @@ static struct snd_kcontrol_new alc262_le > { } /* end */ > }; > > +static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { > + HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_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 = alc262_sony_master_sw_put, > + .private_value = HDA_COMPOSE_AMP_VAL(0x15, 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("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), > + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), > + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), > + { } /* end */ > +}; > + > /* additional init verbs for Benq laptops */ > static struct hda_verb alc262_EAPD_verbs[] = { > {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, > @@ -10169,6 +10189,24 @@ static struct hda_verb alc262_HP_BPC_Wil > { } > }; > > +static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { > + > + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */ > + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, > + {0x14, AC_VERB_SET_CONNECT_SEL, 0x01}, > + > + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* MIC jack */ > + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */ > + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) }, > + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) }, > + > + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP jack */ > + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, > + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, > + {} > +}; > + > + > #ifdef CONFIG_SND_HDA_POWER_SAVE > #define alc262_loopbacks alc880_loopbacks > #endif > @@ -10258,6 +10296,8 @@ static const char *alc262_models[ALC262_ > [ALC262_ULTRA] = "ultra", > [ALC262_LENOVO_3000] = "lenovo-3000", > [ALC262_NEC] = "nec", > + [ALC262_TOSHIBA_S06] = "toshiba-s06", > + [ALC262_TOSHIBA_RX1] = "toshiba-rx1", > [ALC262_AUTO] = "auto", > }; > > @@ -10292,7 +10332,7 @@ static struct snd_pci_quirk alc262_cfg_t > SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), > SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), > SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", > - ALC262_SONY_ASSAMD), > + ALC262_TOSHIBA_RX1), > SND_PCI_QUIRK(0x1179, 0x0268, "Toshiba S06", ALC262_TOSHIBA_S06), > SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), > SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), > @@ -10500,6 +10540,18 @@ static struct alc_config_preset alc262_p > .unsol_event = alc262_toshiba_s06_unsol_event, > .init_hook = alc262_toshiba_s06_init_hook, > }, > + [ALC262_TOSHIBA_RX1] = { > + .mixers = { alc262_toshiba_rx1_mixer }, > + .init_verbs = { alc262_init_verbs, alc262_toshiba_rx1_unsol_verbs }, > + .num_dacs = ARRAY_SIZE(alc262_dac_nids), > + .dac_nids = alc262_dac_nids, > + .hp_nid = 0x03, > + .num_channel_mode = ARRAY_SIZE(alc262_modes), > + .channel_mode = alc262_modes, > + .input_mux = &alc262_capture_source, > + .unsol_event = alc262_hippo_unsol_event, > + .init_hook = alc262_hippo_automute, > + }, > }; > > static int patch_alc262(struct hda_codec *codec) > > -- > HIroshi Miura > NTT DATA Corp. and IPA OSS center > （株）NTTデータ /（独）情報処理推進機構 > 三浦広志 >

1 0

[alsa-devel] [PATCH] ASoC: Convert DAPM pop time configuration to debugfs
by Mark Brown 28 Aug '08

28 Aug '08

Pop time configuration is a debugging feature which shouldn't be used in normal operation so move it to debugfs where it is clearer. This also simplifies the code since debugfs provides standard facilities for exposing simple integer values to user space. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- It would be nice to apply this to 2.6.27 since the sysfs file was only added in the 2.6.27 cycle but it's not critical either way. sound/soc/soc-dapm.c | 53 ++++++++++++++++++------------------------------- 1 files changed, 20 insertions(+), 33 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7a88f76..c016426 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -38,6 +38,7 @@ #include <linux/bitops.h> #include <linux/platform_device.h> #include <linux/jiffies.h> +#include <linux/debugfs.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -67,7 +68,9 @@ static int dapm_status = 1; module_param(dapm_status, int, 0); MODULE_PARM_DESC(dapm_status, "enable DPM sysfs entries"); -static unsigned int pop_time; +static struct dentry *asoc_debugfs; + +static u32 pop_time; static void pop_wait(void) { @@ -817,51 +820,35 @@ static ssize_t dapm_widget_show(struct device *dev, static DEVICE_ATTR(dapm_widget, 0444, dapm_widget_show, NULL); -/* pop/click delay times */ -static ssize_t dapm_pop_time_show(struct device *dev, - struct device_attribute *attr, char *buf) -{ - return sprintf(buf, "%d\n", pop_time); -} - -static ssize_t dapm_pop_time_store(struct device *dev, - struct device_attribute *attr, - const char *buf, size_t count) - -{ - unsigned long val; - - if (strict_strtoul(buf, 10, &val) >= 0) - pop_time = val; - else - printk(KERN_ERR "Unable to parse pop_time setting\n"); - - return count; -} - -static DEVICE_ATTR(dapm_pop_time, 0744, dapm_pop_time_show, - dapm_pop_time_store); - int snd_soc_dapm_sys_add(struct device *dev) { int ret = 0; - if (dapm_status) { - ret = device_create_file(dev, &dev_attr_dapm_widget); + if (!dapm_status) + return 0; - if (ret == 0) - ret = device_create_file(dev, &dev_attr_dapm_pop_time); - } + ret = device_create_file(dev, &dev_attr_dapm_widget); + if (ret != 0) + return ret; - return ret; + asoc_debugfs = debugfs_create_dir("asoc", NULL); + if (!IS_ERR(asoc_debugfs)) + debugfs_create_u32("dapm_pop_time", 0744, asoc_debugfs, + &pop_time); + else + asoc_debugfs = NULL; + + return 0; } static void snd_soc_dapm_sys_remove(struct device *dev) { if (dapm_status) { - device_remove_file(dev, &dev_attr_dapm_pop_time); device_remove_file(dev, &dev_attr_dapm_widget); } + + if (asoc_debugfs) + debugfs_remove_recursive(asoc_debugfs); } /* free all dapm widgets and resources */ -- 1.5.6.5

3 3

[alsa-devel] modifying VIA8237.conf - does it currently work correctly for somebody?
by John L. Utz III 28 Aug '08

28 Aug '08

Hi; I am sorry for the vagueness of the questions, but now that i have some understanding of how these files are supposed to work i want to get this right! to support 4.0/5.1/7.1 for the vt1618 codec i have to change a different set of twiddlers than what is called out in the current version of the conf file. worse yet, the twiddlers called out in the current file dont even *exist* on the vt1618: { name "Surround Down Mix" preserve true value off lock true optional true } { name "Center/LFE Down Mix" preserve true value off lock true optional true } How do i add new stuff without breaking old stuff? is there a sub-identity that i could use? tnx! johnu

2 1