On Jun 18, 2010, at 11:38 AM, alsa-devel-request@xxxxxxxxxxxxxxxx wrote: > Send Alsa-devel mailing list submissions to > alsa-devel@xxxxxxxxxxxxxxxx > > To subscribe or unsubscribe via the World Wide Web, visit > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > or, via email, send a message with subject or body 'help' to > alsa-devel-request@xxxxxxxxxxxxxxxx > > You can reach the person managing the list at > alsa-devel-owner@xxxxxxxxxxxxxxxx > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Alsa-devel digest..." > > > Today's Topics: > > 1. Re: [PATCH 2/5] ALSA: usb-audio: unify UAC macros and struct > names (Daniel Mack) > 2. Re: Timing Info (Clemens Ladisch) > 3. Re: [PATCH] ASoC: pandora: fix CLKX polarity (Mark Brown) > 4. [PATCH] ALSA: hda - add ideapad model for Conexant 5051 codec > (Herton Ronaldo Krzesinski) > 5. Re: [PATCH] ALSA: hda - add ideapad model for Conexant 5051 > codec (Takashi Iwai) > 6. Re: snd_mixer_selem_get_playback_volume return value outside > allowed range (Raymond Yau) > 7. Re: [PATCH] eukrea-tlv320: add support for our i.MX25 board > (Mark Brown) > 8. SOC_DOUBLE_R_SX_TLV (was Update on TLV320AIC3204 Driver) > (Stuart Longland) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 17 Jun 2010 17:26:26 +0200 > From: Daniel Mack <daniel@xxxxxxxx> > Subject: Re: [PATCH 2/5] ALSA: usb-audio: unify UAC > macros and struct names > To: Alex Lee <alexlee188@xxxxxxxxx> > Cc: Takashi Iwai <tiwai@xxxxxxx>, "alsa-devel@xxxxxxxxxxxxxxxx" > <alsa-devel@xxxxxxxxxxxxxxxx>, "clemens@xxxxxxxxxx" > <clemens@xxxxxxxxxx> > Message-ID: <20100617152626.GL17833@xxxxxxxxxxxxxxxxx> > Content-Type: text/plain; charset=us-ascii > > (hint, hint: don't top-post :)) > > On Thu, Jun 17, 2010 at 08:21:22AM -0700, Alex Lee wrote: >> It will not help existing codes, I agree. But it may make writing new UAC2 >> specific code easier? > > Not as long as most of the code stays hybrid for both versions. If there > were versions of these enum values for both UAC1 and UAC2, you would > need to choose one of them in the parser codes. Which one you go for > doesn't actually make any difference as they're both the same, but the > code would read as if it was specific to any version. > > I'd say this would add more to the confusion, and not ease any. > > > Daniel > > > > ------------------------------ > > Message: 2 > Date: Thu, 17 Jun 2010 17:36:59 +0200 > From: Clemens Ladisch <clemens@xxxxxxxxxx> > Subject: Re: Timing Info > To: Paul Dugas <paul@xxxxxxxxxxxxxxxxxxxx> > Cc: alsa-devel@xxxxxxxxxxxxxxxx > Message-ID: <4C1A411B.2060301@xxxxxxxxxx> > Content-Type: text/plain; charset=us-ascii > > Paul Dugas wrote: >> On Sun, Jun 13, 2010 at 10:10 AM, Clemens Ladisch <clemens@xxxxxxxxxx> wrote: >>> Set SND_PCM_TSTAMP_ENABLE to generate a timestamp whenever the hardware >>> position is updated. Call snd_pcm_status to read the position and the >>> corresponding timestamp. >> >> I've re-read this a number of times and have been fiddling with code >> that dumps the status at various points but I'm still not grasping the >> concept here. Can comeone point me to an explanation of the delay, >> avail, and avail_max values? > > The avail value tells you how many frames are available, i.e., can be > written, in the output buffer. The largest value that this had since > the last snd_pcm_status call is avail_max. > > The delay value is the time that will elapse before the next frame that > you write into the buffer will actually be output by the hardware. This > is not only the time until the hardware reads this frame out of the > buffer (which could be computed from avail) but also any processing > after that. > > > HTH > Clemens > > > ------------------------------ > > Message: 3 > Date: Thu, 17 Jun 2010 16:50:01 +0100 > From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx> > Subject: Re: [PATCH] ASoC: pandora: fix CLKX polarity > To: Grazvydas Ignotas <notasas@xxxxxxxxx>, alsa-devel@xxxxxxxxxxxxxxxx > Cc: lrg@xxxxxxxxxxxxxxx > Message-ID: <u6g5khs6pbrq63fngu2ncjvv.1276789801383@xxxxxxxxxxxxxxxxx> > Content-Type: text/plain; charset=utf-8 > > Acked-by: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx> > > Probably best to add a CC to stable too. Sorry about the rubbish formatting, on my phone and it can't do any better. > > Grazvydas Ignotas <notasas@xxxxxxxxx> wrote: > >> After mass production started it was found that several boards exhibit >> noise problems during sound playback. After some investigation it was >> determined that CLKX polarity is set incorrectly, and even if most boards >> can tolerate the wrong setting, there are some that don't. >> >> Fix polarity setup in the board file. As the clock settings for input and >> output now match, merge in and out functions and structures to simplify >> code. >> >> Signed-off-by: Grazvydas Ignotas <notasas@xxxxxxxxx> >> --- >> sound/soc/omap/omap3pandora.c | 36 ++++++++---------------------------- >> 1 files changed, 8 insertions(+), 28 deletions(-) >> >> diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c >> index 87ce842..9eecac1 100644 >> --- a/sound/soc/omap/omap3pandora.c >> +++ b/sound/soc/omap/omap3pandora.c >> @@ -43,12 +43,14 @@ >> >> static struct regulator *omap3pandora_dac_reg; >> >> -static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, >> - struct snd_pcm_hw_params *params, unsigned int fmt) >> +static int omap3pandora_hw_params(struct snd_pcm_substream *substream, >> + struct snd_pcm_hw_params *params) >> { >> struct snd_soc_pcm_runtime *rtd = substream->private_data; >> struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; >> struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; >> + int fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | >> + SND_SOC_DAIFMT_CBS_CFS; >> int ret; >> >> /* Set codec DAI configuration */ >> @@ -91,24 +93,6 @@ static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream, >> return 0; >> } >> >> -static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, >> - struct snd_pcm_hw_params *params) >> -{ >> - return omap3pandora_cmn_hw_params(substream, params, >> - SND_SOC_DAIFMT_I2S | >> - SND_SOC_DAIFMT_IB_NF | >> - SND_SOC_DAIFMT_CBS_CFS); >> -} >> - >> -static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, >> - struct snd_pcm_hw_params *params) >> -{ >> - return omap3pandora_cmn_hw_params(substream, params, >> - SND_SOC_DAIFMT_I2S | >> - SND_SOC_DAIFMT_NB_NF | >> - SND_SOC_DAIFMT_CBS_CFS); >> -} >> - >> static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w, >> struct snd_kcontrol *k, int event) >> { >> @@ -231,12 +215,8 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec) >> return snd_soc_dapm_sync(codec); >> } >> >> -static struct snd_soc_ops omap3pandora_out_ops = { >> - .hw_params = omap3pandora_out_hw_params, >> -}; >> - >> -static struct snd_soc_ops omap3pandora_in_ops = { >> - .hw_params = omap3pandora_in_hw_params, >> +static struct snd_soc_ops omap3pandora_ops = { >> + .hw_params = omap3pandora_hw_params, >> }; >> >> /* Digital audio interface glue - connects codec <--> CPU */ >> @@ -246,14 +226,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { >> .stream_name = "HiFi Out", >> .cpu_dai = &omap_mcbsp_dai[0], >> .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], >> - .ops = &omap3pandora_out_ops, >> + .ops = &omap3pandora_ops, >> .init = omap3pandora_out_init, >> }, { >> .name = "TWL4030", >> .stream_name = "Line/Mic In", >> .cpu_dai = &omap_mcbsp_dai[1], >> .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], >> - .ops = &omap3pandora_in_ops, >> + .ops = &omap3pandora_ops, >> .init = omap3pandora_in_init, >> } >> }; >> -- >> 1.6.3.3 >> > > ------------------------------ > > Message: 4 > Date: Thu, 17 Jun 2010 14:15:06 -0300 > From: Herton Ronaldo Krzesinski <herton@xxxxxxxxxxxxxxx> > Subject: [PATCH] ALSA: hda - add ideapad model for > Conexant 5051 codec > To: alsa-devel@xxxxxxxxxxxxxxxx > Cc: Takashi Iwai <tiwai@xxxxxxx> > Message-ID: <201006171415.07101.herton@xxxxxxxxxxxxxxx> > Content-Type: Text/Plain; charset="us-ascii" > > Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b, > which isn't muted when headphone is plugged in. This adds additional > support to the extra subwoofer via new ideapad model. > > Signed-off-by: Herton Ronaldo Krzesinski <herton@xxxxxxxxxxxxxxx> > --- > Documentation/sound/alsa/HD-Audio-Models.txt | 1 + > sound/pci/hda/patch_conexant.c | 20 ++++++++++++++++++++ > 2 files changed, 21 insertions(+), 0 deletions(-) > > diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt > b/Documentation/sound/alsa/HD-Audio-Models.txt > index 1d38b0d..84e81ad 100644 > --- a/Documentation/sound/alsa/HD-Audio-Models.txt > +++ b/Documentation/sound/alsa/HD-Audio-Models.txt > @@ -282,6 +282,7 @@ Conexant 5051 > hp HP Spartan laptop > hp-dv6736 HP dv6736 > hp-f700 HP Compaq Presario F700 > + ideapad Lenovo IdeaPad laptop > lenovo-x200 Lenovo X200 laptop > toshiba Toshiba Satellite M300 > > diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c > index 2bf2cb5..54f7419 100644 > --- a/sound/pci/hda/patch_conexant.c > +++ b/sound/pci/hda/patch_conexant.c > @@ -1632,6 +1632,11 @@ static void cxt5051_update_speaker(struct hda_codec > *codec) > pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; > snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, > pinctl); > + /* on ideapad there is an aditional speaker (subwoofer) to mute */ > + if (spec->ideapad) > + snd_hda_codec_write(codec, 0x1b, 0, > + AC_VERB_SET_PIN_WIDGET_CONTROL, > + pinctl); > } > > /* turn on/off EAPD (+ mute HP) as a master switch */ > @@ -1888,6 +1893,13 @@ static void cxt5051_init_mic_port(struct hda_codec > *codec, hda_nid_t nid, > #endif > } > > +static struct hda_verb cxt5051_ideapad_init_verbs[] = { > + /* Subwoofer */ > + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, > + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, > + { } /* end */ > +}; > + > /* initialize jack-sensing, too */ > static int cxt5051_init(struct hda_codec *codec) > { > @@ -1917,6 +1929,7 @@ enum { > CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini > Dock 250410 */ > CXT5051_F700, /* HP Compaq Presario F700 */ > CXT5051_TOSHIBA, /* Toshiba M300 & co */ > + CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ > CXT5051_MODELS > }; > > @@ -1927,6 +1940,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { > [CXT5051_LENOVO_X200] = "lenovo-x200", > [CXT5051_F700] = "hp-700", > [CXT5051_TOSHIBA] = "toshiba", > + [CXT5051_IDEAPAD] = "ideapad", > }; > > static struct snd_pci_quirk cxt5051_cfg_tbl[] = { > @@ -1938,6 +1952,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { > CXT5051_LAPTOP), > SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), > SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), > + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), > {} > }; > > @@ -1999,6 +2014,11 @@ static int patch_cxt5051(struct hda_codec *codec) > spec->mixers[0] = cxt5051_toshiba_mixers; > spec->auto_mic = AUTO_MIC_PORTB; > break; > + case CXT5051_IDEAPAD: > + spec->init_verbs[spec->num_init_verbs++] = > + cxt5051_ideapad_init_verbs; > + spec->ideapad = 1; > + break; > } > > return 0; > -- > 1.7.1 > > > > ------------------------------ > > Message: 5 > Date: Thu, 17 Jun 2010 20:39:20 +0200 > From: Takashi Iwai <tiwai@xxxxxxx> > Subject: Re: [PATCH] ALSA: hda - add ideapad model for > Conexant 5051 codec > To: Herton Ronaldo Krzesinski <herton@xxxxxxxxxxxxxxx> > Cc: alsa-devel@xxxxxxxxxxxxxxxx > Message-ID: <s5hk4pxld0n.wl%tiwai@xxxxxxx> > Content-Type: text/plain; charset=US-ASCII > > At Thu, 17 Jun 2010 14:15:06 -0300, > Herton Ronaldo Krzesinski wrote: >> >> Lenovo IdeaPad Y430 has an additional subwoofer connected at pin 0x1b, >> which isn't muted when headphone is plugged in. This adds additional >> support to the extra subwoofer via new ideapad model. >> >> Signed-off-by: Herton Ronaldo Krzesinski <herton@xxxxxxxxxxxxxxx> > > Thanks, applied now. > > > Takashi > >> --- >> Documentation/sound/alsa/HD-Audio-Models.txt | 1 + >> sound/pci/hda/patch_conexant.c | 20 ++++++++++++++++++++ >> 2 files changed, 21 insertions(+), 0 deletions(-) >> >> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt >> b/Documentation/sound/alsa/HD-Audio-Models.txt >> index 1d38b0d..84e81ad 100644 >> --- a/Documentation/sound/alsa/HD-Audio-Models.txt >> +++ b/Documentation/sound/alsa/HD-Audio-Models.txt >> @@ -282,6 +282,7 @@ Conexant 5051 >> hp HP Spartan laptop >> hp-dv6736 HP dv6736 >> hp-f700 HP Compaq Presario F700 >> + ideapad Lenovo IdeaPad laptop >> lenovo-x200 Lenovo X200 laptop >> toshiba Toshiba Satellite M300 >> >> diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c >> index 2bf2cb5..54f7419 100644 >> --- a/sound/pci/hda/patch_conexant.c >> +++ b/sound/pci/hda/patch_conexant.c >> @@ -1632,6 +1632,11 @@ static void cxt5051_update_speaker(struct hda_codec >> *codec) >> pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0; >> snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, >> pinctl); >> + /* on ideapad there is an aditional speaker (subwoofer) to mute */ >> + if (spec->ideapad) >> + snd_hda_codec_write(codec, 0x1b, 0, >> + AC_VERB_SET_PIN_WIDGET_CONTROL, >> + pinctl); >> } >> >> /* turn on/off EAPD (+ mute HP) as a master switch */ >> @@ -1888,6 +1893,13 @@ static void cxt5051_init_mic_port(struct hda_codec >> *codec, hda_nid_t nid, >> #endif >> } >> >> +static struct hda_verb cxt5051_ideapad_init_verbs[] = { >> + /* Subwoofer */ >> + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, >> + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, >> + { } /* end */ >> +}; >> + >> /* initialize jack-sensing, too */ >> static int cxt5051_init(struct hda_codec *codec) >> { >> @@ -1917,6 +1929,7 @@ enum { >> CXT5051_LENOVO_X200, /* Lenovo X200 laptop, also used for Advanced Mini >> Dock 250410 */ >> CXT5051_F700, /* HP Compaq Presario F700 */ >> CXT5051_TOSHIBA, /* Toshiba M300 & co */ >> + CXT5051_IDEAPAD, /* Lenovo IdeaPad Y430 */ >> CXT5051_MODELS >> }; >> >> @@ -1927,6 +1940,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { >> [CXT5051_LENOVO_X200] = "lenovo-x200", >> [CXT5051_F700] = "hp-700", >> [CXT5051_TOSHIBA] = "toshiba", >> + [CXT5051_IDEAPAD] = "ideapad", >> }; >> >> static struct snd_pci_quirk cxt5051_cfg_tbl[] = { >> @@ -1938,6 +1952,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { >> CXT5051_LAPTOP), >> SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), >> SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), >> + SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo IdeaPad", CXT5051_IDEAPAD), >> {} >> }; >> >> @@ -1999,6 +2014,11 @@ static int patch_cxt5051(struct hda_codec *codec) >> spec->mixers[0] = cxt5051_toshiba_mixers; >> spec->auto_mic = AUTO_MIC_PORTB; >> break; >> + case CXT5051_IDEAPAD: >> + spec->init_verbs[spec->num_init_verbs++] = >> + cxt5051_ideapad_init_verbs; >> + spec->ideapad = 1; >> + break; >> } >> >> return 0; >> -- >> 1.7.1 >> > > > ------------------------------ > > Message: 6 > Date: Fri, 18 Jun 2010 07:39:07 +0800 > From: Raymond Yau <superquad.vortex2@xxxxxxxxx> > Subject: Re: snd_mixer_selem_get_playback_volume return > value outside allowed range > To: ALSA Development Mailing List <alsa-devel@xxxxxxxxxxxxxxxx> > Message-ID: > <AANLkTin7y2JzkaCYb4p69uDfWOo4cTa85in0Q3l7RwbD@xxxxxxxxxxxxxx> > Content-Type: text/plain; charset=ISO-8859-1 > > 2010/6/17 Clemens Ladisch <clemens@xxxxxxxxxx> > >> Raymond Yau wrote: >>> Should snd_mixer_selem_get_playback_volume() perform range check ? >> >> No, it's the responsibility of the control implementation to return >> valid values (and to check that values that are being set are valid). >> >> >> Regards, >> Clemens >> > > If it is the responsibility of the driver/external plugin to keep the value > inside the allowed range > > your patch actually help those buggy driver/external plugin hide the bug > only > > > ------------------------------ > > Message: 7 > Date: Fri, 18 Jun 2010 01:56:37 +0100 > From: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx> > Subject: Re: [PATCH] eukrea-tlv320: add support for our > i.MX25 board > To: Eric B?nard <eric@xxxxxxxxxx> > Cc: alsa-devel@xxxxxxxxxxxxxxxx, s.hauer@xxxxxxxxxxxxxx, > lrg@xxxxxxxxxxxxxxx > Message-ID: <20100618005637.GA3333@xxxxxxxxxxxxxxxxxxxxxxxxxxx> > Content-Type: text/plain; charset=iso-8859-1 > > On Thu, Jun 17, 2010 at 03:44:01PM +0200, Eric B?nard wrote: >> * tdm slot has to be configured to get sound working on i.MX25 >> >> Signed-off-by: Eric B?nard <eric@xxxxxxxxxx> > > Applied but I had to hand apply the Kconfig update, please check that it > applied correctly. > > > ------------------------------ > > Message: 8 > Date: Fri, 18 Jun 2010 12:38:10 +1000 > From: Stuart Longland <redhatter@xxxxxxxxxx> > Subject: SOC_DOUBLE_R_SX_TLV (was Update on TLV320AIC3204 > Driver) > To: Mark Brown <broonie@xxxxxxxxxxxxxxxxxxxxxxxxxxx> > Cc: Takashi Iwai <tiwai@xxxxxxx>, alsa-devel@xxxxxxxxxxxxxxxx, Eric > B??nard <eric@xxxxxxxxxx>, Liam Girdwood <lrg@xxxxxxxxxxxxxxx> > Message-ID: <20100618023810.GS7759@xxxxxxxxxxxxxxxxxxxxxxx> > Content-Type: text/plain; charset=us-ascii > > On Tue, Jun 15, 2010 at 03:11:10PM +1000, Stuart Longland wrote: >> When a read is performed, this structure points to this function as the >> means for reading the register (defined in sound/soc-core.c): >> >> /** >> * snd_soc_get_volsw_2r_sx - double with tlv and variable data size >> * mixer get callback >> * @kcontrol: mixer control >> * @uinfo: control element information >> * >> * Returns 0 for success. >> */ >> int snd_soc_get_volsw_2r_sx(struct snd_kcontrol *kcontrol, >> struct snd_ctl_elem_value *ucontrol) >> { >> struct soc_mixer_control *mc = >> (struct soc_mixer_control *)kcontrol->private_value; >> struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); >> unsigned int mask = (1<<mc->shift)-1; >> int min = mc->min; >> int val = snd_soc_read(codec, mc->reg) & mask; >> int valr = snd_soc_read(codec, mc->rreg) & mask; >> >> ucontrol->value.integer.value[0] = ((val & 0xff)-min); >> ucontrol->value.integer.value[1] = ((valr & 0xff)-min); >> return 0; >> } >> EXPORT_SYMBOL_GPL(snd_soc_get_volsw_2r_sx); > > I've figured it out now... and there'll be a patch to fix it. > > The mute <--> gain interaction was pure coincidence as it turns out. > There's no interaction between the two, just that when I toggle the > mute, the gain gets re-read, and hence it *appeared* to interact. > > Notice the value calculated for the channel is: > > output = ( regval & 0xff ) - minimum > > In my case, my minimum is -6. Suppose I set my driver gain to -1dB, in > hexadecimal, 0x3f. Since no sign extension is done, (regval & 0xff) = > 63. Adding 6 to this (double negation) you get 69... which is greater > than the maximum gain of 29. Hence the scale shoots up to maximum. > > The fix, is to change the above so that they mask the result; thus > wrapping around the zero point like so: > > output = ( ( regval & 0xff ) - minimum ) & mask > > This fixes the problem that I observe, and shouldn't break the other > user of the control. A patch to fix this issue is coming. > -- > Stuart Longland (aka Redhatter, VK4MSL) .'''. > Gentoo Linux/MIPS Cobalt and Docs Developer '.'` : > . . . . . . . . . . . . . . . . . . . . . . .'.' > http://dev.gentoo.org/~redhatter :.' > > I haven't lost my mind... > ...it's backed up on a tape somewhere. > > > ------------------------------ > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel@xxxxxxxxxxxxxxxx > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > End of Alsa-devel Digest, Vol 40, Issue 80 > ****************************************** _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
