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[alsa-devel] Driver - PCM MMAP mode timestamps
by Jaroslav Kysela 08 Jan '08

08 Jan '08

Hi, it seems that we have another inconsistency with PCM MMAP mode timestamps. If I read code correctly, snd_pcm_status() returns timestamp from last interrupt BUT updates hw_ptr to current (calling snd_pcm_update_hw_ptr() again). It means, that application will get unrelated timestamp to hw position returned in snd_pcm_status() in case of mmap mode. I don't see any way to use this timestamp now. To fix this issue, I would like to create a next status ioctl which will return only mmap timestamp plus last related interrupt ring buffer position and remove 'if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_MMAP)' condition from snd_pcm_status() in driver completely. Any opinion? Other idea? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] ALC268 "test" model
by Jonathan Woithe 08 Jan '08

08 Jan '08

Hi Tobin > > I try to use a patch from J. Woithe to put a test model into the > > ALC268 driver [1] (enabling all pins etc.); unfortunately it gives a > > lot of compiling errors ([2]) which I am unable to understand as I > > don't have sufficient knowledge of the internal API of HDA codec. > > > > Can someone please look over it and correct the mistakes? > > > > Marco Schuster > > > > [1]: http://harddisk.is-a-geek.org/patch.txt > > [2]: http://harddisk.is-a-geek.org/compiling-errors.txt > > Hi, and thanks for the effort. I'll fix this up a bit and send out an > updated patch this weekend. I have an ALC268 test board to work with > now, so I should be able to get it figured out. I'm not entirely sure what's going on here with this patch. As is indicated by the patch itself it's against ALSA in the 2.6.24-rc6 kernel and it applies and compiles fine for me. The last few error messages at the second URL seems to suggest that the patch was in some way corrupted because I'm seeing non-ASCII characters in the error messages. I should note that the patch as referenced by Marco was not believed by me to be ready for release yet - that is why I haven't forwarded it on to the list for consideration yet. I sent it to Marco privately and I had hoped to utilise Marco's system to verify basic functionality of the patch before posting it for consideration. It was prepared in a hurry at the end of last year just before I went on leave and have not had a chance to revisit the patch or the issues that Marco saw until now. Since you have an ALC268 board it's probably sensible for you to take over the patch at this point - I don't have an ALC268 system. However, understand that by making this initial patch available to Marco privately it was not my intention to create a lot of extra work for yourself (or any other developer for that matter). I had intended to see it through to operational status before bothering the list with it. Are you happy to take over management of this patch or would you like me to resume dialog with Marco? If the patch at the above URL is in some way corrupted drop me a line and I'll forward you the original. Regards jonathan

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[alsa-devel] [PATCH] hda: Dynamically create digital gain mixers
by Matthew Ranostay 08 Jan '08

08 Jan '08

Dynamically create digital gain mixers for dmics that have out-amp support. Also some 92HD73xx's codecs don't have DMIC gains, so this also prevents creating dead mixers. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 37956a45462d pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jan 07 15:16:37 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 07 15:04:42 2008 -0500 @@ -463,9 +463,6 @@ static struct hda_verb stac92hd73xx_6ch_ /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -490,9 +487,6 @@ static struct hda_verb stac92hd73xx_8ch_ /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -519,9 +513,6 @@ static struct hda_verb stac92hd73xx_10ch /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -635,10 +626,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -665,10 +652,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -695,10 +678,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -725,10 +704,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), @@ -745,10 +720,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -2313,15 +2284,18 @@ static int stac92xx_auto_create_dmic_inp struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *dimux = &spec->private_dimux; hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; - int i, j; + int err, i, j; + char name[32]; dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0]; dimux->items[dimux->num_items].index = 0; dimux->num_items++; for (i = 0; i < spec->num_dmics; i++) { + hda_nid_t nid; int index; int num_cons; + unsigned int wcaps; unsigned int def_conf; def_conf = snd_hda_codec_read(codec, @@ -2332,17 +2306,32 @@ static int stac92xx_auto_create_dmic_inp if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; + nid = spec->dmic_nids[i]; num_cons = snd_hda_get_connections(codec, spec->dmux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); for (j = 0; j < num_cons; j++) - if (con_lst[j] == spec->dmic_nids[i]) { + if (con_lst[j] == nid) { index = j; goto found; } continue; found: + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + sprintf(name, "%s Capture Volume", + stac92xx_dmic_labels[dimux->num_items]); + + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_VOL, + name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + dimux->items[dimux->num_items].label = stac92xx_dmic_labels[dimux->num_items]; dimux->items[dimux->num_items].index = index;

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[alsa-devel] HDA: Enable chipset gcap usage
by Tobin Davis 08 Jan '08

08 Jan '08

Summary: hda_intel.c use gcap register to determine number of streams supported by southbridge. This patch removes hardcoded values for the number of streams supported by the southbridge in most chipsets, and reads these values from the chipset directly. Most systems are hardwired for 4 streams in each direction, but newer chipsets change that capability. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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Re: [alsa-devel] fujitsu siemens amilo la 1703 vt1807 hda intel card problem
by Tobin Davis 07 Jan '08

07 Jan '08

The information you provided is good, we're just a bit behind. I'm trying to get some programming specs for this codec now. In the mean time, could you try using a tip build of the alsa driver? You can download it here: ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver/ This may have some newer code that will enable your audio. Also, try the drivers on http://viaarena.com. THey may have an updated version that hasn't hit the main tree yet. Tobin On Mon, 2008-01-07 at 21:59 +0100, Norayr Chilingaryan wrote: > Excuse me for bothering you. > May be I need to provide other info? > Did you receive the link? > Di you have any ideas? > > Thank you > > Norayr > > > On Jan 5, 2008 5:53 PM, Norayr Chilingaryan < gnorayr(a)gmail.com> > wrote: > > http://pastebin.ca/842745 > > Thank you very much! > > > > > > On Jan 5, 2008 3:56 AM, Tobin Davis <tdavis(a)dsl-only.net> > wrote: > > Could you download and run > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > > This script will provide a lot of useful information. > After running it, please reply to this email with the > link that the script generates, so we can retrieve the > data, thanks. > > Tobin > > > > > On Sat, 2008-01-05 at 02:48 +0100, Norayr Chilingaryan > wrote: > > > > Hello I decided to write here, firtly, because my > > messages in aksa-user have been ingored, and > > secondly, because I understood that my card most > > probably just unsupported and I must contact > > developers. I have tried to pass options > > model=fujitsu, because I have a fujitsu siements > > notebook etc I have also tried to pass amny other > > options in /etc/modules.d/alsa but withnot effect I > > use gentoo, and the only thing I noticed with latest > > kernels and alsa drivers 1.0.15 is that I can hear > > sound through the headphones, but with low volume. > > It is not possible to get volume higher or speaker > > to make sounds now. So, please, this is information > > about the card 04:01.0 Audio device: VIA > > Technologies, Inc. VIA High Definition Audio > > Controller (rev 10) kakraphoon ~ # > > cat /proc/asound/card0/codec#* | grep Codec Codec: > > VIA VIA VT1708 Codec: Motorola Si3054 Thank you very > > much Norayr > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- Tobin Davis > > > Hain't we got all the fools in town on our side? And hain't that a big > enough majority in any town? > -- Mark Twain, "Huckleberry Finn" > > > > > -- Tobin Davis Anything anybody can say about America is true. -- Emmett Grogan

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[alsa-devel] Cross compiling alsa-lib (again)
by Adrian McMenamin 07 Jan '08

07 Jan '08

Well, I searched through 96 weeks of archived alsa-devel postings to get to this one: http://article.gmane.org/gmane.linux.alsa.devel/37120 (Which shows I am doing the right thing in any case.) This post recommends this: CFLAGS="-Os -m4 -ml" ./configure --prefix=/usr --host=sh4-unknown-linux-gnu --target=sh4-unknown-linux-gnu --build=i686-pc-linux-gnu I admit to not using the additional cflags stuff before, but even if I use this: CFLAGS="-Os -m4 -ml" ./configure --host=sh4-linux --prefix=/usr CC=/home/adrian/dreamy-linux/buildroot/build_sh4/staging_dir/bin/sh4-linux-gcc --disable-python --disable-rawmidi --enable-static --with-confdir=/home/adrian/dreamy-linux/buildroot/project_build_sh4/dreamcast/root/usr/share/alsa --disable-mixer --enable-static=yes --enable-shared=no --target=sh4-linux --build=i686-linux Followed by: make and make DESTDIR=/home/adrian/dreamy-linux/buildroot/project_build_sh4/dreamcast/root install I still get this: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: AICA [AICA], device 0: AICA PCM [AICA PCM] Subdevices: 1/1 Subdevice #0: subdevice #0 Followed by: $ aplay -L null Discard all samples (playback) or generate zero samples (capture) $ aplay /aine-email.wav Playing WAVE '/aine-email.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' aplay: set_params:895: Access type not available $ aplay -D hw:0,0 /aine-email.wav Playing WAVE '/aine-email.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: set_params:895: Access type not available Just as before. Any clues?

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Re: [alsa-devel] [PATCH] hda: Advanced power management for codecs.
by Takashi Iwai 07 Jan '08

07 Jan '08

At Mon, 07 Jan 2008 12:31:49 -0500, Matthew Ranostay wrote: > > >> @@ -2830,7 +2883,12 @@ static void stac92xx_hp_detect(struct hd > >> for (i = 0; i < cfg->speaker_outs; i++) > >> stac92xx_set_pinctl(codec, cfg->speaker_pins[i], > >> AC_PINCTL_OUT_EN); > >> - } > >> + for (i = 0; i < cfg->hp_outs; i++) > >> + pwr_state |= 1 << (cfg->hp_pins[i] - 0xa); > >> + } > >> + > >> + /* power down unused ports */ > >> + snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, pwr_state); > >> > > > > Is this verb available on every STAC codec? I took a look at some > > specs but they don't show this verb at all. > > > Ah no this a new one for recent codecs, probably need to some sort of > check for this. Or is it equivalent to power down each unused pin widget? If so, we can do it somehow better in the common code. Takashi

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[alsa-devel] [PATCH] hda: Advanced power management for codecs.
by Matthew Ranostay 07 Jan '08

07 Jan '08

Added support for advanced power management features in new IDT codecs. Inactive ADCs and DACs are changed to D3 mode when not in playback or capture, also inactive line-out/headphone ports are disabled when jack detect does not sense a connection. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 5b03c0176aa5 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jan 07 13:33:45 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 07 11:22:56 2008 -0500 @@ -160,6 +160,7 @@ struct sigmatel_spec { /* i/o switches */ unsigned int io_switch[2]; + unsigned int io_state[2]; unsigned int clfe_swap; unsigned int aloopback; @@ -438,6 +439,52 @@ static int stac92xx_aloopback_put(struct kcontrol->private_value >> 16, dac_mode); return 1; +} + +static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) +{ + int i; + + for (i = 0; i < spec->multiout.num_dacs; i++) { + if (spec->multiout.dac_nids[i] == nid) + return 1; + } + + return 0; +} + +static int stac92xx_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = kcontrol->private_value & 0xffff; + int idx = (kcontrol->private_value >> 19) & 0xf; + int chs = (kcontrol->private_value >> 16) & 0x3; + long *valp = ucontrol->value.integer.value; + unsigned int wid_caps, val, change; + + change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!is_in_dac_nids(spec, nid)) + nid = spec->adc_nids[idx]; + + val = *valp++; + wid_caps = get_wcaps(codec, nid); + if (wid_caps & AC_WCAP_POWER) { + if (chs == 3) + val |= *valp; + else { /* Center/LFE mixers workaround */ + spec->io_state[--chs] = val; + val |= spec->io_state[chs ? 0 : 1]; + } + + /* set to power state D3 mode if DAC/ADC isn't being used */ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, + val ? 0: AC_PWRST_D3); + } +#endif + return change; } static struct hda_verb stac9200_core_init[] = { @@ -622,6 +669,20 @@ static struct hda_verb stac9205_core_ini .private_value = verb_read | (verb_write << 16), \ } +#define STAC_CODEC_MUTE_IDX(xname, xcidx, nid, xindex, direction) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xcidx, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = stac92xx_mixer_amp_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, xindex, direction), \ + } + +#define STAC_CODEC_MUTE(xname, nid, xindex, direction) \ + STAC_CODEC_MUTE_IDX(xname, 0, nid, xindex, direction) + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -640,10 +701,10 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x1, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -670,10 +731,10 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x1, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -700,10 +761,10 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x1, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -730,12 +791,12 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x1, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x1, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), @@ -751,12 +812,12 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x1, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x1, HDA_OUTPUT), { } /* end */ }; @@ -857,7 +918,7 @@ static int stac92xx_build_controls(struc if (err < 0) return err; } - return 0; + return 0; } static unsigned int ref9200_pin_configs[8] = { @@ -1962,7 +2023,7 @@ enum { static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), - HDA_CODEC_MUTE(NULL, 0, 0, 0), + STAC_CODEC_MUTE(NULL, 0, 0, 0), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2059,19 +2120,6 @@ static int stac92xx_add_dyn_out_pins(str return 0; } - -static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - - for (i = 0; i < spec->multiout.num_dacs; i++) { - if (spec->multiout.dac_nids[i] == nid) - return 1; - } - - return 0; -} - /* * Fill in the dac_nids table from the parsed pin configuration * This function only works when every pin in line_out_pins[] @@ -2806,6 +2854,7 @@ static void stac92xx_hp_detect(struct hd struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int i, presence; + unsigned char pwr_state = 0; presence = 0; for (i = 0; i < cfg->hp_outs; i++) { @@ -2816,12 +2865,16 @@ static void stac92xx_hp_detect(struct hd if (presence) { /* disable lineouts, enable hp */ - for (i = 0; i < cfg->line_outs; i++) + for (i = 0; i < cfg->line_outs; i++) { stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); - for (i = 0; i < cfg->speaker_outs; i++) + pwr_state |= 1 << (cfg->line_out_pins[i] - 0xa); + } + for (i = 0; i < cfg->speaker_outs; i++) { stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + pwr_state |= 1 << (cfg->speaker_pins[i] - 0xa); + } } else { /* enable lineouts, disable hp */ for (i = 0; i < cfg->line_outs; i++) @@ -2830,7 +2883,12 @@ static void stac92xx_hp_detect(struct hd for (i = 0; i < cfg->speaker_outs; i++) stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); - } + for (i = 0; i < cfg->hp_outs; i++) + pwr_state |= 1 << (cfg->hp_pins[i] - 0xa); + } + + /* power down unused ports */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, pwr_state); } static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)

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[alsa-devel] snd_pcm_status_get_tstamp()
by Lennart Poettering 07 Jan '08

07 Jan '08

Hi! The documentation for snd_pcm_status_get_tstamp() is not really enlightening. After a call to snd_pcm_status() what data does the tstamp field actually carry? Some kind of system clock timestamp when the data in the status struct was valid, or some kind of sound card clock timestamp that is relative to the point in time where the device was opened/started? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Status of snd_pcm_rewind()
by Lennart Poettering 07 Jan '08

07 Jan '08

Hi! What's the status of snd_pcm_rewind()? Is it available on all devices? I assume not (at least dmix doesn't...). But I guess I can safely assume it is available when mmap is used? Is there any API function to test if it is available? I am working on adding "glitch-free" support to PulseAudio. I.e. have a huge playback buffer (> 2s), and schedule playback based on system timers instead of sound card interrupts, thus allowing us to dynamically adjust the inetrrupt frequency to the clients that are connected, and thus save power. I can enable this technique only if I can rewrite the playback buffer whenever I need to, thus I need to in advance if snd_pcm_rewind() will work or not. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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