November 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Fix internal mic on Dell Vostro 3500
by Julian Wollrath 09 Nov '11

09 Nov '11

Hello, the internal mic of my Dell Vostro 3500 was not working with kernel v3.2-rc1 (and propably with none of the older kernels). By choosing dell-s14 as the model I was able to get the internal mic working, but now my externel mic jack was a second headphone out jack. So I looked at the differences between the pin configurations of the auto model and the dell- s14 model and found out, that setting the value of on 0x11 from default 0x400000f6 to 0x90a60160 was responsible for the working internal mic. I attached a patch that fixes this problem but it has a flaw. Since I did not know what value I have to put in the second argument of SND_PCI_QUIRK I put 0x0000 as dummy in there. It works for me but if anyone could give me a hint on how to obtain, what I really have to put in there, I would be happy to fill in the right value. With best regards Julian Wollrath

2 3

[alsa-devel] [PATCH v3] ASoC: wm9081: Use snd_soc_update_bits for read-modify-write
by Axel Lin 09 Nov '11

09 Nov '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/wm9081.c | 82 ++++++++++++++++++++------------------------- 1 files changed, 36 insertions(+), 46 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index b491ae1..f032078 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -826,84 +826,75 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { static int wm9081_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* VMID=2*40k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x2; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x2); /* Normal bias current */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, 0); break; case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg &= ~WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, 0); /* Select startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC | WM9081_BIAS_ENA); /* VMID 2*4k; Soft VMID ramp enable */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg |= WM9081_VMID_RAMP | 0x6; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | + WM9081_VMID_SEL_MASK, + WM9081_VMID_RAMP | 0x6); mdelay(100); /* Normal bias enable & soft start off */ - reg &= ~WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | + WM9081_VMID_SEL_MASK, 0); /* Standard bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, 0); } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x04; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x04); /* Standby bias current on */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, + WM9081_STBY_BIAS_ENA); break; case SND_SOC_BIAS_OFF: /* Startup bias source and disable bias */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC; - reg &= ~WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC); /* Disable VMID with soft ramping */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_VMID_SEL_MASK, + WM9081_VMID_RAMP); /* Actively discharge LINEOUT */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg |= WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, + WM9081_LINEOUT_DISCH); break; } @@ -1291,11 +1282,10 @@ static int wm9081_probe(struct snd_soc_codec *codec) wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ - reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT); - snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); - reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); - snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, - reg | WM9081_SPKPGAZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, + WM9081_LINEOUTZC, WM9081_LINEOUTZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, + WM9081_SPKPGAZC, WM9081_SPKPGAZC); if (!wm9081->pdata.num_retune_configs) { dev_dbg(codec->dev, -- 1.7.5.4

1 2

[alsa-devel] Spdif: Control interface
by ajay khandelwal 09 Nov '11

09 Nov '11

Hi All, On my PC SPDIF control is exposing mute and volume control. I understand for mute but how volume control is exported for digital audio. Best Regards, Ajay

3 5

Re: [alsa-devel] Why my linux OS has only 8 sound cards ? How can I modify it ?
by Takashi Iwai 09 Nov '11

09 Nov '11

[Please stop using HTML mail for posting to a ML...] At Wed, 9 Nov 2011 10:24:41 +0800, 沈发旺 wrote: > > Hi,Takashi Iwai : > * I used your patch ,but it also display this when I used the command > "ls /dev/dsp*".* > */dev/dsp /dev/dsp2 /dev/dsp4 /dev/dsp6 > /dev/dsp1 /dev/dsp3 /dev/dsp5 /dev/dsp7* > *It also display this when board up:* > *input: C-Media USB Headphone Set as > /devices/platform/fsl-ehci.1/usb2/2-1/2-1.5/2-1.5:1.3/input/input7 > generic-usb 0003:0D8C:000C.0007: input: USB HID v1.00 Device [C-Media USB > Headphone Set ] on usb-fsl-ehci.1-1.5/input3 > usb 2-1.6: new full speed USB device using fsl-ehci and address 8 > unable to register OSS PCM device 8:0 > unable to register OSS mixer device 8:0 > input: C-Media USB Headphone Set as > /devices/platform/fsl-ehci.1/usb2/2-1/2-1.6/2-1.6:1.3/input/input8 > generic-usb 0003:0D8C:000C.0008: input: USB HID v1.00 Device [C-Media USB > Headphone Set ] on usb-fsl-ehci.1-1.6/input3 > usb 2-1.7: new full speed USB device using fsl-ehci and address 9 > unable to register OSS PCM device 9:0 > unable to register OSS mixer device 9:0 > input: C-Media USB Headphone Set as > /devices/platform/fsl-ehci.1/usb2/2-1/2-1.7/2-1.7:1.3/input/input9 > generic-usb 0003:0D8C:000C.0009: input: USB HID v1.00 Device [C-Media USB > Headphone Set ] on usb-fsl-ehci.1-1.7/input3 > **when the sound cards exceed 8 ,I found it return err from > sound/sound_core.c:__sound_insert_unit( ) . I read this code and found > modify the function of "register_sound_special_device " -> max_unit = 128 + > chain; when I change to :** max_unit = 256 + chain, my board will display > 10 sound cards :* So it's a limitation in sound_core.c. (BTW, 256+chain is wrong. It should be simply 256.) I vaguely remember why we left the limit 8 in ALSA side. One reason was that sound_core itself doesn't support more than 8, and we weren't sure whether changing it would regress or not. For example, don't apps expect only a single digit after /dev/dsp? OTOH, having 11 sound devices with OSS must be a rare case and I'm sure a legacy system using OSS won't have such a configuration. And fixing the code is basically a right thing. That being said, I'm fine to change that part for the upstream, if no one objects. thanks, Takashi > /dev/dsp /dev/dsp2 /dev/dsp4 /dev/dsp6 /dev/dsp8 > /dev/dsp1 /dev/dsp3 /dev/dsp5 /dev/dsp7 /dev/dsp9 > *Thank you vary mach !* > 2011/11/9 Takashi Iwai <tiwai(a)suse.de> > > > At Tue, 08 Nov 2011 18:37:53 +0100, > > Clemens Ladisch wrote: > > > > > > Takashi Iwai wrote: > > > > Linus Torvalds wrote: > > > > > Takashi, any idea? Do we have some 8-sound-card limit somewhere? > > > > > > > > > > I do see that we have that odd > > > > > > > > > > if (card && card->number >= 8) > > > > > return 0; /* ignore silently */ > > > > > > > > > > in sound/core/sound_oss.c: snd_register_oss_device(), but it has no > > > > > comments. The thing goes back to 2005 with the commit log entry of > > > > > "[ALSA] dynamic minors (6/6): increase maximum number of sound > > cards", > > > > > but the value '8'isn't explained. > > > > > > > > ALSA itself can support 32 cards when CONFIG_SND_DYNAMIC_MINORS is > > > > set ... > > > > The above code is, however, about OSS-emulation. And I guess it's > > > > just a left-over of the old static limitation. > > > > > > Yes; I didn't change the limit in the OSS emulation because the minor > > > number layout doesn't have space for all 32 cards, and extending the > > > limit to 16 did not seem worth the effort for a 'legacy' interface. > > > > > > > Actually OSS could support up to 16 cards (in theory). > > > > > > > > Below is an untested fix patch. Does it work? > > > > > > Not without increasing SNDRV_OSS_MINORS, too: > > > > Ah, good catch. Below is the revised patch. > > > > > > Takashi > > > > --- > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: Fix the card number limit of OSS-emulation > > > > There are left-over codes from the ancient days with the static device > > number limitation of 8. Actaully OSS can support up to 16 cards. > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > v2: fix SNDRV_OSS_MINORS, too > > > > sound/core/sound_oss.c | 6 +++--- > > 1 files changed, 3 insertions(+), 3 deletions(-) > > > > diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c > > index c700920..e952833 100644 > > --- a/sound/core/sound_oss.c > > +++ b/sound/core/sound_oss.c > > @@ -35,7 +35,7 @@ > > #include <linux/sound.h> > > #include <linux/mutex.h> > > > > -#define SNDRV_OSS_MINORS 128 > > +#define SNDRV_OSS_MINORS 256 > > > > static struct snd_minor *snd_oss_minors[SNDRV_OSS_MINORS]; > > static DEFINE_MUTEX(sound_oss_mutex); > > @@ -111,7 +111,7 @@ int snd_register_oss_device(int type, struct snd_card > > *card, int dev, > > int register1 = -1, register2 = -1; > > struct device *carddev = snd_card_get_device_link(card); > > > > - if (card && card->number >= 8) > > + if (card && card->number >= SNDRV_MINOR_OSS_DEVICES) > > return 0; /* ignore silently */ > > if (minor < 0) > > return minor; > > @@ -170,7 +170,7 @@ int snd_unregister_oss_device(int type, struct > > snd_card *card, int dev) > > int track2 = -1; > > struct snd_minor *mptr; > > > > - if (card && card->number >= 8) > > + if (card && card->number >= SNDRV_MINOR_OSS_DEVICES) > > return 0; > > if (minor < 0) > > return minor; > > -- > > 1.7.7 > > > >

1 0

[alsa-devel] soc-dsp questions
by Vinod Koul 09 Nov '11

09 Nov '11

Hey Liam, I had few questions on your dsp work, would be great if you could answer them and help me in understanding the approach. - Is it possible to have one FE connected to multiple BEs (voice FE to modem and headset bBE), if so how? - SND_SOC_DAI_LINK_OPT_HOST, what does this mean, does not seem to be used anywhere in core? - Why comment need dummy codec for some BEs like BT etc, when soc-core has already one? - any reason why be haves a number associated with a BE, than usual convention of referring things by string name in asoc? and last one, when would this be merged to mainline, 3.3? -- ~Vinod

2 2

[alsa-devel] [PATCH] ALSA: HDA: Remove quirk for Toshiba T110
by David Henningsson 09 Nov '11

09 Nov '11

According to the bug reporter, model=auto is needed to make the internal microphone work. BugLink: https://bugs.launchpad.net/bugs/819699 Reported-by: Andrej (agno01) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e706e4..0de2119 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3062,7 +3062,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1993, "Asus U50F", CXT5066_ASUS), SND_PCI_QUIRK(0x1179, 0xff1e, "Toshiba Satellite C650D", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5), - SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5066_LAPTOP), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT5066_OLPC_XO_1_5), -- 1.7.5.4

2 1

[alsa-devel] [RFC] hda - is_jack_detectable()
by Raymond Yau 09 Nov '11

09 Nov '11

Refer to HDA Specification 7.3.3.31 Configuration Default Misc[3:0] is a bit field used to indicate other information about the jack. Currently, only bit 0 is defined. If bit 0 is set, it indicates that the jack has no presence detect capability, so even if a Pin Complex indicates that the codec hardware supports the presence detect functionality on the jack, the external circuitry is not capable of supporting the functionality. The bit definitions for the Miscfield are in Table 106. Do we need to change is_jack_detectable() when user setup BIOS to use AC97 Front Audio Panel which set AC_DEFCFG_MISC_NO_PRESENCE bit ? +#define get_defcfg_misc(cfg) \ + ((cfg & AC_DEFCFG_MISC) >> AC_DEFCFG_MISC_SHIFT) static inline bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid) { return (snd_hda_query_pin_caps(codec, nid) & AC_PINCAP_PRES_DETECT) && + (!(get_defcfg_misc(codec,nid) & AC_DEFCFG_MISC_NO_PRESENCE)) && (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP); } or add another fuction is_defcfg_misc_no_presence() to indicate jack has no presence detect capability ?

3 10

[alsa-devel] the monitors with working Displayport audio
by Wu Fengguang 09 Nov '11

09 Nov '11

Hi Takashi, I enumerated all the DELL DisplayPort monitors in our lab and find only two of them being able to produce audible DisplayPort audio... - DELL 2408WFP - DELL 2709W - DELL 3008WFP no CEA extension found - DELL U2410 ELD reports OK but cannot hear DP sound monitor_present 1 eld_valid 1 monitor_name DELL U2410 connection_type DisplayPort eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xac10 product_id 0xf017 port_id 0x0 support_hdcp 0 support_ai 0 audio_sync_delay 0 speakers [0x1] FL/FR sad_count 1 sad0_coding_type [0x1] LPCM sad0_channels 2 sad0_rates [0xe0] 32000 44100 48000 sad0_bits [0xe0000] 16 20 24 - DELL U3011 ELD and DP sound both OK monitor_present 1 eld_valid 1 monitor_name DELL U3011 connection_type DisplayPort eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xac10 product_id 0x4065 port_id 0x0 support_hdcp 0 support_ai 0 audio_sync_delay 0 speakers [0xf] FL/FR LFE FC RL/RR sad_count 1 sad0_coding_type [0x1] LPCM sad0_channels 6 sad0_rates [0x1ee0] 32000 44100 48000 96000 176400 192000 384000 sad0_bits [0xe0000] 16 20 24 - Eizo FlexScan EV2333W ELD and DP sound both OK monitor_present 1 eld_valid 1 monitor_name EV2333W connection_type DisplayPort eld_version [0x2] CEA-861D or below edid_version [0x3] CEA-861-B, C or D manufacture_id 0xc315 product_id 0x2068 port_id 0x0 support_hdcp 0 support_ai 1 audio_sync_delay 0 speakers [0x1] FL/FR sad_count 1 sad0_coding_type [0x1] LPCM sad0_channels 2 sad0_rates [0x1ee0] 32000 44100 48000 96000 176400 192000 384000 sad0_bits [0xe0000] 16 20 24 Thanks, Fengguang

2 2

Re: [alsa-devel] Why my linux OS has only 8 sound cards ? How can I modify it ?
by Linus Torvalds 08 Nov '11

08 Nov '11

Chris - you need to pick a better set of people to ask. A mailing list, right maintainer etc. Takashi, any idea? Do we have some 8-sound-card limit somewhere? I do see that we have that odd if (card && card->number >= 8) return 0; /* ignore silently */ in sound/core/sound_oss.c: snd_register_oss_device(), but it has no comments. The thing goes back to 2005 with the commit log entry of "[ALSA] dynamic minors (6/6): increase maximum number of sound cards", but the value '8'isn't explained. Linus On Tue, Nov 8, 2011 at 12:30 AM, Chris Shen <chris2shen(a)gmail.com> wrote: > Hi,everyone! > My version of kernel is linux-2.6.35 .And my board has 10 sounds > cards.After system was up,I used the comand "aplay -l ".Then console > would display this: > **** List of PLAYBACK Hardware Devices **** > card 0: imx3stack [imx-3stack], device 0: WM8731 WM8731-0 [] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 1: default [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 2: default_1 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 3: default_2 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 4: default_3 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 5: default_4 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 6: default_5 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 7: default_6 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 8: default_7 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > card 9: default_8 [C-Media USB Headphone Set ], device 0: USB Audio > [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > But,when I used the command "ls /dev/dsp*" ,the console display: > /dev/dsp /dev/dsp2 /dev/dsp4 /dev/dsp6 > /dev/dsp1 /dev/dsp3 /dev/dsp5 /dev/dsp7 > when the board is up ,I found have this err "unable to register OSS > mixer device 8:0" and "unable to register OSS mixer device 9:0" . > I think the kernel's max sound cards is 8 .How can I modify it ? > Thank you! >

3 3

[alsa-devel] [regression] probably in mixer.c code
by Alexey Fisher 08 Nov '11

08 Nov '11

Hallo all, Takashi, there is one regression in current linux main tree: cval->res quirks do not work any more. See usb/mixer.c:1150 Are there was some work in this area or may be some one bisected this bug down? Regards, Alexey.

2 10