March 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Intel: Add suspend_pre and resume_post for Braswell snd_soc_card
by Jin Yao 13 Mar '15

13 Mar '15

On Braswell, we need to add some machine specific setting before suspend and after resume. For example, disable/enable jack detection in codec so use snd_soc_card suspend_pre and resume_post ops for this purpose. Signed-off-by: Jin Yao <yao.jin(a)linux.intel.com> --- sound/soc/intel/cht_bsw_rt5672.c | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/sound/soc/intel/cht_bsw_rt5672.c b/sound/soc/intel/cht_bsw_rt5672.c index 279df4c..c41fae3 100644 --- a/sound/soc/intel/cht_bsw_rt5672.c +++ b/sound/soc/intel/cht_bsw_rt5672.c @@ -267,6 +267,35 @@ static struct snd_soc_dai_link cht_dailink[] = { }, }; +static int cht_suspend_pre(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + dev_dbg(codec->dev, "disabling jack detect before going to suspend.\n"); + rt5670_jack_suspend(codec); + break; + } + } + return 0; +} + +static int cht_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec; + + list_for_each_entry(codec, &card->codec_dev_list, card_list) { + if (!strcmp(codec->component.name, "i2c-10EC5670:00")) { + dev_dbg(codec->dev, "enabling jack detect for resume.\n"); + rt5670_jack_resume(codec); + break; + } + } + + return 0; +} + /* SoC card */ static struct snd_soc_card snd_soc_card_cht = { .name = "cherrytrailcraudio", @@ -278,6 +307,8 @@ static struct snd_soc_card snd_soc_card_cht = { .num_dapm_routes = ARRAY_SIZE(cht_audio_map), .controls = cht_mc_controls, .num_controls = ARRAY_SIZE(cht_mc_controls), + .suspend_pre = cht_suspend_pre, + .resume_post = cht_resume_post, }; static int snd_cht_mc_probe(struct platform_device *pdev) -- 1.9.1

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[alsa-devel] Issues w/ Creative Labs [SB X-Fi Xtreme Audio] CA0110-IBG
by Alnie 13 Mar '15

13 Mar '15

At Wed, 25 Feb 2015 00:21:15 -0800, Alnie wrote: > > > > On 02/25/2015 12:06 AM, Takashi Iwai wrote: > > At Tue, 24 Feb 2015 23:55:25 -0800, > > Alnie wrote: > >> > >> [ 2619.199658] pciehp 0000:00:1c.3:pcie04: Card not present on Slot(3) > >> [ 2619.199666] pciehp 0000:00:1c.3:pcie04: slot(3): Link Down event > >> [ 2619.199674] pciehp 0000:00:1c.3:pcie04: Link Down event ignored on > >> slot(3): already powering off > >> [ 2619.199911] pci_bus 0000:06: busn_res: [bus 06-0c] is released > >> [ 2620.757928] pciehp 0000:00:1c.3:pcie04: Card present on Slot(3) > >> [ 2620.767950] pciehp 0000:00:1c.3:pcie04: slot(3): Link Up event > >> [ 2620.767985] pciehp 0000:00:1c.3:pcie04: Link Up event ignored on > >> slot(3): already powering on > > > > pciehp reacted power off and on here, and... > > > >> [ 2620.876160] pci 0000:05:00.0: [1102:7006] type 01 class 0x060400 > >> [ 2620.876426] pci 0000:05:00.0: supports D1 D2 > >> [ 2620.876719] pci 0000:05:00.0: bridge configuration invalid ([bus > >> 00-00]), reconfiguring > >> [ 2620.876928] pci_bus 0000:06: busn_res: can not insert [bus 06-ff] > >> under [bus 05-0c] (conflicts with (null) [bus 05-0c]) > > > > .... and something failed here for the PCI bridge device. > > > >> [ 2620.876967] pci 0000:06:00.0: [1102:0009] type 00 class 0x040300 > >> [ 2620.877013] pci 0000:06:00.0: reg 0x10: [mem 0x00000000-0x00003fff] > >> [ 2620.877203] pci 0000:06:00.0: supports D1 D2 > >> [ 2620.877443] pci 0000:05:00.0: PCI bridge to [bus 06-ff] > >> [ 2620.877462] pci 0000:05:00.0: bridge window [io 0x0000-0x0fff] > >> [ 2620.877474] pci 0000:05:00.0: bridge window [mem 0x00000000-0x000fffff] > >> [ 2620.877492] pci 0000:05:00.0: bridge window [mem > >> 0x00000000-0x000fffff 64bit pref] > >> [ 2620.877500] pci_bus 0000:06: busn_res: [bus 06-ff] end is updated to 06 > >> [ 2620.877559] pci 0000:05:00.0: BAR 14: assigned [mem > >> 0xf4300000-0xf43fffff] > >> [ 2620.877568] pci 0000:05:00.0: BAR 15: assigned [mem > >> 0xf4000000-0xf40fffff 64bit pref] > >> [ 2620.877574] pci 0000:05:00.0: BAR 13: assigned [io 0x2000-0x2fff] > >> [ 2620.877582] pci 0000:06:00.0: BAR 0: assigned [mem 0xf4300000-0xf4303fff] > >> [ 2620.877595] pci 0000:05:00.0: PCI bridge to [bus 06] > >> [ 2620.877603] pci 0000:05:00.0: bridge window [io 0x2000-0x2fff] > >> [ 2620.877618] pci 0000:05:00.0: bridge window [mem 0xf4300000-0xf43fffff] > >> [ 2620.877630] pci 0000:05:00.0: bridge window [mem > >> 0xf4000000-0xf40fffff 64bit pref] > >> [ 2620.877647] pcieport 0000:00:1c.3: PCI bridge to [bus 05-0c] > >> [ 2620.877655] pcieport 0000:00:1c.3: bridge window [io 0x2000-0x2fff] > >> [ 2620.877667] pcieport 0000:00:1c.3: bridge window [mem > >> 0xf4300000-0xf7efffff] > >> [ 2620.877677] pcieport 0000:00:1c.3: bridge window [mem > >> 0xf4000000-0xf40fffff 64bit pref] > >> [ 2620.877990] pci 0000:05:00.0: enabling device (0000 -> 0003) > >> [ 2620.878013] snd_hda_intel 0000:06:00.0: enabling device (0000 -> 0002) > >> [ 2620.882747] azx_single_wait_for_response: 108 callbacks suppressed > >> [ 2620.882753] snd_hda_intel 0000:06:00.0: Codec #1 probe error; > >> disabling it... > >> [ 2620.887052] snd_hda_intel 0000:06:00.0: no AFG or MFG node found > >> [ 2620.887058] snd_hda_intel 0000:06:00.0: no codecs initialized > > > > What if you once unload snd-hda-intel module and reload again while > > keeping the card plugged? I guess it still doesn't work, but just to > > be sure... > > > > Overall, this look more like a problem of PCI core stuff. Better to > > toss the issue to PCI people. > > > > > > Takashi > > > > I did alsa force-reload and the same error comes up. Do you know of a > good starting point for presenting PCI issues? > >Ask on linux-pci at vger.kernel.org ? > > >Takashi Bjorn has suggested it is not a PCI issue... http://article.gmane.org/gmane.linux.kernel.pci/39492 Any other ideas in regards to this? -Alnie

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[alsa-devel] [PATCH v2] ALSA: control: Add sanity checks for user ctl id name string
by Takashi Iwai 13 Mar '15

13 Mar '15

There was no check about the id string of user control elements, so we accepted even a control element with an empty string, which is obviously bogus. This patch adds more sanity checks of id strings. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: fix typos, thinkos sound/core/control.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/control.c b/sound/core/control.c index 35324a8e83c8..eeb691d1911f 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1170,6 +1170,10 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, if (info->count < 1) return -EINVAL; + if (!*info->id.name) + return -EINVAL; + if (strnlen(info->id.name, sizeof(info->id.name)) >= sizeof(info->id.name)) + return -EINVAL; access = info->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (info->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| -- 2.3.2

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[alsa-devel] [PATCH] ALSA: control: Fix breakage of user ctl element addition
by Takashi Iwai 13 Mar '15

13 Mar '15

In the commit [2225e79b9b03: 'ALSA: core: reduce stack usage related to snd_ctl_new()'], the id field of the newly added kctl is untouched, thus all attribute like name string remain empty. The fix is just to add the forgotten memcpy of the id field. Fixes: 2225e79b9b03 ('ALSA: core: reduce stack usage related to snd_ctl_new()') Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/control.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/control.c b/sound/core/control.c index 54a412af3224..d677c27746e9 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1267,6 +1267,7 @@ static int snd_ctl_elem_add(struct snd_ctl_file *file, err = snd_ctl_new(&kctl, count, access, file); if (err < 0) return err; + memcpy(&kctl->id, &info->id, sizeof(kctl->id)); kctl->private_data = kzalloc(sizeof(struct user_element) + private_size, GFP_KERNEL); if (kctl->private_data == NULL) { -- 2.3.2

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[alsa-devel] Microphone Not Working after Suspend on Apple MacBookAir5, 2 Cirrus Logic CS4206
by Vasil Zlatanov 12 Mar '15

12 Mar '15

Issue as seen on reddit thread: https://www.reddit.com/r/linux/comments/1kccbk/alsa_bug_driving_me_crazy_pl… A common problem for Apple Macbook Air 5.2 users where microphone does not work after system resume. As users have noted microphone starts working ones again when the volume is changed in alsa-mixer. alsa-info dumps are attached showing before and after being adjusted (where before causes microphone not to work while in after it does) IRC user 'debianuser' kindly helped out and found that using the two following commands fix the issue: amixer -cPCH set "Capture" 0% nocap amixer -cPCH set "Capture" 100% cap In my experience both are required and they seem to have the same effect as manually changing the volume level in alsamixer, and as such resolve the issue (until another suspend when they have to be run again). As of now I'm resolving this issue by creating a service that is to run after resume from susspend with the given amixer commands to renable the microphone. The second issue, which is also in the reddit thread, is that the microphine is mono and when recording only right channel audio is seen. For me this broke programs such as mumble and skype. A alsa-info output with coefficients is attached. 'debianuser' managed to find a solution to this issue well, resolving it by running the following two commands: hda-verb /dev/snd/hwC0D0 0x11 SET_COEF_INDEX 2 hda-verb /dev/snd/hwC0D0 0x11 SET_PROC_COEF 0x102a After one runs them a single time after boot, the channels collectly show and programs issues are resolved. 'debianuser' mentions that the sources for the commands are available in the kernel and applied to the macbook 8.1 model, but not to mine.

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Re: [alsa-devel] Microphone Not Working after Suspend on Apple MacBookAir5, 2 Cirrus Logic CS4206
by Takashi Iwai 12 Mar '15

12 Mar '15

At Thu, 12 Mar 2015 20:13:05 +0100, Vasil Zlatanov wrote: > > Applied both patches against 3.18.6-1 and correctly resolved all > issues. Good to hear, I queued both patches now. They'll be backported to stable kernels once when merged to upstream. thanks, Takashi > > On Thu, Mar 12, 2015 at 11:38:39AM +0100, Takashi Iwai wrote: > > At Wed, 11 Mar 2015 23:26:00 +0100, > > Vasil Zlatanov wrote: > > > > > > Issue as seen on reddit thread: > > > https://www.reddit.com/r/linux/comments/1kccbk/alsa_bug_driving_me_crazy_pl… > > > > > > A common problem for Apple Macbook Air 5.2 users where microphone does > > > not work after system resume. As users have noted microphone starts > > > working ones again when the volume is changed in alsa-mixer. > > > > > > alsa-info dumps are attached showing before and after being adjusted > > > (where before causes microphone not to work while in after it does) > > > > > > IRC user 'debianuser' kindly helped out and found that using the two > > > following commands fix the issue: > > > > > > amixer -cPCH set "Capture" 0% nocap > > > amixer -cPCH set "Capture" 100% cap > > > > > > In my experience both are required and they seem to have the same effect > > > as manually changing the volume level in alsamixer, and as such resolve > > > the issue (until another suspend when they have to be run again). > > > > This might be due to the single ADC amp. The ADC widgets have > > multiple inputs and usually they have individual amp volumes. But > > some codecs give only a single volume for all inputs. > > > > There is a flag in the driver to change the control for that. The > > patch below enables it. > > > > diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c > > index 1589c9bcce3e..dd2b3d92071f 100644 > > --- a/sound/pci/hda/patch_cirrus.c > > +++ b/sound/pci/hda/patch_cirrus.c > > @@ -584,6 +584,7 @@ static int patch_cs420x(struct hda_codec *codec) > > return -ENOMEM; > > > > spec->gen.automute_hook = cs_automute; > > + codec->single_adc_amp = 1; > > > > snd_hda_pick_fixup(codec, cs420x_models, cs420x_fixup_tbl, > > cs420x_fixups); > > > > Alternatively, you can pass "single_adc_smp = true" hint in the > > firmware patch passed via patch option. See > > Documentation/sound/alsa/HD-Audio.txt for details. > > > > > > > As of now I'm resolving this issue by creating a service that is to run > > > after resume from susspend with the given amixer commands to renable the > > > microphone. > > > > > > The second issue, which is also in the reddit thread, is that the > > > microphine is mono and when recording only right channel audio is seen. > > > For me this broke programs such as mumble and skype. > > > > > > A alsa-info output with coefficients is attached. > > > > > > 'debianuser' managed to find a solution to this issue well, resolving it > > > by running the following two commands: > > > > > > hda-verb /dev/snd/hwC0D0 0x11 SET_COEF_INDEX 2 > > > hda-verb /dev/snd/hwC0D0 0x11 SET_PROC_COEF 0x102a > > > > > > After one runs them a single time after boot, the channels collectly > > > show and programs issues are resolved. 'debianuser' mentions that the > > > sources for the commands are available in the kernel and applied to the > > > macbook 8.1 model, but not to mine. > > > > Yes, it's the very same workaround for MBP8,1. You can pass > > model=mbp81 option. If it works, you can just add a sline to > > cs420x_fixup_tbl[] something like > > > > @@ -393,6 +393,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { > > SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), > > SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), > > SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), > > + SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81), > > SND_PCI_QUIRK(0x106b, 0x5b00, "MacBookAir 4,2", CS420X_MBA42), > > SND_PCI_QUIRK_VENDOR(0x106b, "Apple", CS420X_APPLE), > > {} /* terminator */ > > > > > > > > Takashi > [2 <application/pgp-signature (7bit)>] >

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[alsa-devel] [PATCH] ASoC: rt5645: Add TDM support for rt5650
by Bard Liao 12 Mar '15

12 Mar '15

rt5650 and rt5645 use different register bits for TDM configuration. This patch modifies rt5645_set_tdm_slot to support both codecs. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 41 ++++++++++++++++++++++++++++++----------- 1 file changed, 30 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index debf16c..af468ae 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2283,23 +2283,42 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; - unsigned int val = 0; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int i_slot_sft, o_slot_sft, i_width_sht, o_width_sht, en_sft; + unsigned int mask, val = 0; + switch (rt5645->codec_type) { + case CODEC_TYPE_RT5650: + en_sft = 15; + i_slot_sft = 10; + o_slot_sft = 8; + i_width_sht = 6; + o_width_sht = 4; + mask = 0x8ff0; + break; + default: + en_sft = 14; + i_slot_sft = o_slot_sft = 12; + i_width_sht = o_width_sht = 10; + mask = 0x7c00; + break; + } if (rx_mask || tx_mask) { - val |= (1 << 14); - snd_soc_update_bits(codec, RT5645_BASS_BACK, - RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); + val |= (1 << en_sft); + if (rt5645->codec_type == CODEC_TYPE_RT5645) + snd_soc_update_bits(codec, RT5645_BASS_BACK, + RT5645_G_BB_BST_MASK, RT5645_G_BB_BST_25DB); } switch (slots) { case 4: - val |= (1 << 12); + val |= (1 << i_slot_sft) | (1 << o_slot_sft); break; case 6: - val |= (2 << 12); + val |= (2 << i_slot_sft) | (2 << o_slot_sft); break; case 8: - val |= (3 << 12); + val |= (3 << i_slot_sft) | (3 << o_slot_sft); break; case 2: default: @@ -2308,20 +2327,20 @@ static int rt5645_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, switch (slot_width) { case 20: - val |= (1 << 10); + val |= (1 << i_width_sht) | (1 << o_width_sht); break; case 24: - val |= (2 << 10); + val |= (2 << i_width_sht) | (2 << o_width_sht); break; case 32: - val |= (3 << 10); + val |= (3 << i_width_sht) | (3 << o_width_sht); break; case 16: default: break; } - snd_soc_update_bits(codec, RT5645_TDM_CTRL_1, 0x7c00, val); + snd_soc_update_bits(codec, RT5645_TDM_CTRL_1, mask, val); return 0; } -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v3 0/3] add sound effect module waves
by han.lu＠intel.com 12 Mar '15

12 Mar '15

From: "Lu, Han" <han.lu(a)intel.com> Merged 5 patches(0004 to 0008) of last revision into 3 new patches Lu, Han (3): ASoC: Intel: add kcontrol to enable/disable sound effect module waves ASoC: Intel: add function to set parameter to sound effect module waves ASoC: Intel: add kcontrol to set parameter to sound effect module waves sound/soc/intel/sst-haswell-ipc.c | 142 ++++++++++++++++++++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 35 ++++++++++ sound/soc/intel/sst-haswell-pcm.c | 118 +++++++++++++++++++++++++++++++ 3 files changed, 295 insertions(+) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: wm8994: Manage wm8994's MCLK in codec driver
by Inha Song 12 Mar '15

12 Mar '15

Previously, When we use wm8994 codec, we should control its MCLK in machine driver. But, It should be managed by wm8994 codec driver, not machine driver. This patch add MCLK clock DT parsing logic and control own MCLK. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- sound/soc/codecs/wm8994.c | 27 ++++++++++++++++++++++++++- sound/soc/codecs/wm8994.h | 1 + 2 files changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 247b390..d80dcc6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -16,6 +16,7 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/clk.h> #include <linux/gcd.h> #include <linux/i2c.h> #include <linux/platform_device.h> @@ -2470,6 +2471,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = wm8994->wm8994; + int i; wm_hubs_set_bias_level(codec, level); @@ -2491,8 +2493,19 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; } - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) { + if (!IS_ERR(wm8994->clk_mclk[i])) + clk_prepare_enable(wm8994->clk_mclk[i]); + } active_reference(codec); + } else if (codec->dapm.bias_level == SND_SOC_BIAS_ON) { + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) { + if (!IS_ERR(wm8994->clk_mclk[i])) + clk_disable_unprepare( + wm8994->clk_mclk[i]); + } + } break; case SND_SOC_BIAS_STANDBY: @@ -4451,6 +4464,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8994 = { static int wm8994_probe(struct platform_device *pdev) { struct wm8994_priv *wm8994; + int i; wm8994 = devm_kzalloc(&pdev->dev, sizeof(struct wm8994_priv), GFP_KERNEL); @@ -4460,6 +4474,17 @@ static int wm8994_probe(struct platform_device *pdev) wm8994->wm8994 = dev_get_drvdata(pdev->dev.parent); + for (i = 0; i < ARRAY_SIZE(wm8994->clk_mclk); i++) { + char mclk_name[] = "MCLKx"; + + snprintf(mclk_name, sizeof(mclk_name), "MCLK%d", i + 1); + + wm8994->clk_mclk[i] = devm_clk_get(pdev->dev.parent, mclk_name); + if (IS_ERR(wm8994->clk_mclk[i])) + if (PTR_ERR(wm8994->clk_mclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 6536f8d..8cf06a4 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -78,6 +78,7 @@ struct wm8994_priv { int sysclk[2]; int sysclk_rate[2]; int mclk[2]; + struct clk *clk_mclk[2]; int aifclk[2]; int aifdiv[2]; int channels[2]; -- 2.0.0.390.gcb682f8

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[alsa-devel] ti, codec property in sound/soc/omap/omap-twl4030.c not used
by Belisko Marek 12 Mar '15

12 Mar '15

Hi, I'm looking at sources for omap-twl4030.c driver which is used by many boards. In DT bindings documentation is property ti,codec described as mandatory but it's never parsed in driver code. I did small experiment in gta04 board and remove ti,codec from devicetree and sound was working. I think it can be removed from documentation and also from various device trees. Is my assumption correct? Thanks. BR, marek -- as simple and primitive as possible ------------------------------------------------- Marek Belisko - OPEN-NANDRA Freelance Developer Ruska Nova Ves 219 | Presov, 08005 Slovak Republic Tel: +421 915 052 184 skype: marekwhite twitter: #opennandra web: http://open-nandra.com

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