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

[alsa-devel] Alsa 1.0.23 problems with VIA VT1708S
by William King 18 Jun '10

18 Jun '10

A few different issues with this system, but for devel and debugging I have 3 laptops of the exact same hardware. Asus k50AB laptops with the VIA VT1708S sound card. I am documenting as best as I can on this ubuntu bug ticket: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/595606 Currently the issues are: 1. Front mic does not work. No audio with any tests, even arecord test. 2. headphone jack does not work. When headphones are plugged in, the sound continues to come out of the speakers, and no sound out of the headphones. I am able to test patches easily. I have just installed from source 1.0.23 release.

1 0

Re: [alsa-devel] [Alsa-user] Crash 2.6.32.x, Echo 3G, Alsa, SCSI
by Giuliano Pochini 18 Jun '10

18 Jun '10

On Thu, 17 Jun 2010 08:11:06 -0400 "F. Heitkamp" <heitkamp(a)ameritech.net> wrote: > >> My goal is to be able to plug in a MIDI keyboard, or USB guitar > >> interface and play along with songs etc. > >> > >> I plugged in a MIDI keyboard, "/proc/asound/cards" showed the keyboard > >> was recognized. The keyboard has a USB and MIDI interface. > >> I am not sure if they both work at the same time. I had the MIDI cable > >> plugged into the Gina3G, but I could not tell if the keyboard was > >> recognized. > >> I compiled csound5. When I engaged it the system immediately froze and > >> started spitting out: > >> [...] > > Ok, I compiled csound, but I don't know how to use it. How can I reproduce > > the problem ? > > > Sorry I took so long to get back to you. > It turns out Csound is not related to the problem. > > In this case I am only using kernel 2.6.33.5-rt23-amd64, alsa lib 1.0.2, > but 2.6.34 does it too. > I bring up firefox. Web pages attempt to play sound. The kernel > begans doing: > > [ 2879.766331] allocate_pipes: ch=1 int=2 > [ 2879.766332] allocate_pipes: channel already open > [ 2879.766333] allocate_pipes(1) err=-11 > [ 2879.766334] pcm_hw_freed > [ 2.821190] pcm_analog_out_open cs=0 oc=2 r=44100 > [ 2879.821266] allocate_pipes: ch=1 int=2 > [ 2879.821267] allocate_pipes: channel already open > > ,infinitely. The system is unusable at this point and I have to > Alt-SysRq-k to get back in. > > Let me know what other information to provide. Perhaps is it related with the OSS emulation ? Do you know if the browser plays the sound itself or if it is using an external program or plugin ? -- Giuliano.

1 0

[alsa-devel] Cannot find
by subapriya sakthivel 18 Jun '10

18 Jun '10

Hay guys, I have download eclair branch code using below commands. repo init -u git://android.git.kernel.org/platform/manifest.git -b eclair repo sync But inside Hardware folder, i cannot find alsa_sound folder. Can anyone help me on this? Am i chekout from correct branch? If so, where can i get this folder. Thanks for your help in advance. -Priya

3 2

[PATCH] soc/imx: clean Kconfig
by Eric Bénard 18 Jun '10

18 Jun '10

Signed-off-by: Eric Bénard <eric(a)eukrea.com> --- sound/soc/imx/Kconfig | 10 ++-------- 1 files changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 6ef57e0..35e6cf6 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -8,22 +8,17 @@ config SND_IMX_SOC Say Y or M if you want to add support for codecs attached to the i.MX SSI interface. -config SND_MXC_SOC_SSI - tristate - config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" depends on SND_IMX_SOC && MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 - select SND_MXC_SOC_SSI help Enable support for audio on the i.MX31ADS with the WM1133-EV1 PMIC board with WM8835x fitted. config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" - depends on MACH_PCM043 || MACH_PCA100 - select SND_MXC_SOC_SSI + depends on SND_IMX_SOC && (MACH_PCM043 || MACH_PCA100) select SND_SOC_WM9712 help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -31,8 +26,7 @@ config SND_SOC_PHYCORE_AC97 config SND_SOC_EUKREA_TLV320 bool "Eukrea TLV320" - depends on MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD - select SND_IMX_SOC + depends on SND_IMX_SOC && (MACH_EUKREA_MBIMX27_BASEBOARD || MACH_EUKREA_MBIMXSD_BASEBOARD) select SND_SOC_TLV320AIC23 help Enable I2S based access to the TLV320AIC23B codec attached -- 1.6.3.3 --------------020001010209060103040907--

1 0

Re: [alsa-devel] Contents of Alsa-devel digest...
by Junhwa Rhee 18 Jun '10

18 Jun '10

On Jun 18, 2010, at 11:38 AM, alsa-devel-request(a)alsa-project.org wrote: > Send Alsa-devel mailing list submissions to > alsa-devel(a)alsa-project.org > > 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(a)alsa-project.org > > You can reach the person managing the list at > alsa-devel-owner(a)alsa-project.org > > 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(a)caiaq.de> > Subject: Re: [alsa-devel] [PATCH 2/5] ALSA: usb-audio: unify UAC > macros and struct names > To: Alex Lee <alexlee188(a)gmail.com> > Cc: Takashi Iwai <tiwai(a)suse.de>, "alsa-devel(a)alsa-project.org" > <alsa-devel(a)alsa-project.org>, "clemens(a)ladisch.de" > <clemens(a)ladisch.de> > Message-ID: <20100617152626.GL17833(a)buzzloop.caiaq.de> > 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(a)ladisch.de> > Subject: Re: [alsa-devel] Timing Info > To: Paul Dugas <paul(a)dugasenterprises.com> > Cc: alsa-devel(a)alsa-project.org > Message-ID: <4C1A411B.2060301(a)ladisch.de> > Content-Type: text/plain; charset=us-ascii > > Paul Dugas wrote: >> On Sun, Jun 13, 2010 at 10:10 AM, Clemens Ladisch <clemens(a)ladisch.de> 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(a)opensource.wolfsonmicro.com> > Subject: Re: [alsa-devel] [PATCH] ASoC: pandora: fix CLKX polarity > To: Grazvydas Ignotas <notasas(a)gmail.com>, alsa-devel(a)alsa-project.org > Cc: lrg(a)slimlogic.co.uk > Message-ID: <u6g5khs6pbrq63fngu2ncjvv.1276789801383(a)email.android.com> > Content-Type: text/plain; charset=utf-8 > > Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > 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(a)gmail.com> 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(a)gmail.com> >> --- >> 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(a)mandriva.com.br> > Subject: [alsa-devel] [PATCH] ALSA: hda - add ideapad model for > Conexant 5051 codec > To: alsa-devel(a)alsa-project.org > Cc: Takashi Iwai <tiwai(a)suse.de> > Message-ID: <201006171415.07101.herton(a)mandriva.com.br> > 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(a)mandriva.com.br> > --- > 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(a)suse.de> > Subject: Re: [alsa-devel] [PATCH] ALSA: hda - add ideapad model for > Conexant 5051 codec > To: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> > Cc: alsa-devel(a)alsa-project.org > Message-ID: <s5hk4pxld0n.wl%tiwai(a)suse.de> > 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(a)mandriva.com.br> > > 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(a)gmail.com> > Subject: Re: [alsa-devel] snd_mixer_selem_get_playback_volume return > value outside allowed range > To: ALSA Development Mailing List <alsa-devel(a)alsa-project.org> > Message-ID: > <AANLkTin7y2JzkaCYb4p69uDfWOo4cTa85in0Q3l7RwbD(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > 2010/6/17 Clemens Ladisch <clemens(a)ladisch.de> > >> 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(a)opensource.wolfsonmicro.com> > Subject: Re: [alsa-devel] [PATCH] eukrea-tlv320: add support for our > i.MX25 board > To: Eric B?nard <eric(a)eukrea.com> > Cc: alsa-devel(a)alsa-project.org, s.hauer(a)pengutronix.de, > lrg(a)slimlogic.co.uk > Message-ID: <20100618005637.GA3333(a)opensource.wolfsonmicro.com> > 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(a)eukrea.com> > > 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(a)gentoo.org> > Subject: [alsa-devel] SOC_DOUBLE_R_SX_TLV (was Update on TLV320AIC3204 > Driver) > To: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > Cc: Takashi Iwai <tiwai(a)suse.de>, alsa-devel(a)alsa-project.org, Eric > B??nard <eric(a)eukrea.com>, Liam Girdwood <lrg(a)slimlogic.co.uk> > Message-ID: <20100618023810.GS7759(a)www.longlandclan.yi.org> > 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(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > End of Alsa-devel Digest, Vol 40, Issue 80 > ******************************************

1 0

[alsa-devel] [PATCH] ALSA: hda - add ideapad model for Conexant 5051 codec
by Herton Ronaldo Krzesinski 17 Jun '10

17 Jun '10

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(a)mandriva.com.br> --- 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

2 1

Re: [alsa-devel] [PATCH] ASoC: pandora: fix CLKX polarity
by Mark Brown 17 Jun '10

17 Jun '10

Acked-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> 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(a)gmail.com> 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(a)gmail.com> >--- > 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 >

1 0

[alsa-devel] Timing Info
by Paul Dugas 17 Jun '10

17 Jun '10

I'm using snd_pcm_readi() to collect samples and am wondering if/how I can get accurate timestamps for them. Seems like the snd_pcm_status() gives me this info but I'm unclear on the specifics. Can someone point me toward the correct field in the status structure that I should be using and what sample that timestamp corresponds to? Does snd_pcm_status_get_tstamp() get me the timestamp for the first frame returned by the next snd_pcm_readi() call? P

2 6

Re: [alsa-devel] [PATCH 2/5] ALSA: usb-audio: unify UAC macros and struct names
by Daniel Mack 17 Jun '10

17 Jun '10

(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

1 0

[alsa-devel] error code 130: io_open - ALSA open error
by S. Aguinaga 17 Jun '10

17 Jun '10

Hello Folks, I tried finding what this error code means, but my searches come up empty. Would any of you point me to the location where these error codes are described or shoot me an email with a brief explanation. // Sal Aguinaga // Northwestern University

2 1