January 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] replace logical- by bit-and in alc_subsystem_id(), sound/pci/hda/patch_realtek.c
by Roel Kluin 28 Jan '08

28 Jan '08

This was found in linus' git tree, am I right here? -- replace logical- by bit-and Signed-off-by: Roel Kluin <12o3l(a)tiscali.nl> --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1c50278..0a13c53 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -739,7 +739,7 @@ static void alc_subsystem_id(struct hda_codec *codec, /* check sum */ tmp = 0; for (i = 1; i < 16; i++) { - if ((ass >> i) && 1) + if ((ass >> i) & 1) tmp++; } if (((ass >> 16) & 0xf) != tmp)

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[alsa-devel] A Weird issue on Gigabyte board
by Jiang zhe 28 Jan '08

28 Jan '08

Hi, I meet with a weird issue. Please refer to the bug 0003674 on bugtrack.alsa-project.org. It use alc889 codec but the codec vendor id is 0885. With the alsa-driver-hg20080110.tar.bz2, it works well with option "model=6stack-dig". But after I simply added the this line into the alc882_cfg_tbl[]: SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35-DS3R", ALC882_6ST_DIG), The dmesg still show "hda_codec: Unknown model for ALC882, trying auto-probe from BIOS...". and driver will crash. With this patch, even loading it with option "model=6stack-dig" doesn't help, driver will still crash. I found a line in the dmesg: "hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0xbe0f0009." The Cad is 0xb. The codec_mask can change from 0x4 to 0x1: alsa-kernel/pci/hda/hda_intel.c:732: codec_mask = 0x4 alsa-kernel/pci/hda/hda_intel.c:732: codec_mask = 0x1 The os is kubuntu,kernel version is 2.6.22-14. Here is some dmesg info: [ 36.934143] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:732: codec_mask = 0x4 [ 36.935975] hda-codec: In the pacth_alc882,board_config is -1 [ 36.935976] hda_codec: Unknown model for ALC882, trying auto-probe from BIOS... [ 36.936469] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2794: autoconfig: line_outs=4 (0x14/0x15/0x16/0x17/0x0) [ 36.936471] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2798: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 36.936473] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2802: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) [ 36.936476] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2810: inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x0, aux=0x0 [ 37.935491] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:581: hda_intel: azx_get_response timeout, switching to polling mode: last cmd=0xe1cf0009 [ 38.936857] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:588: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0xe1cf0009 [ 38.936937] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:644: hda-intel: get_response timeout: IRS=0x1 [ 38.936940] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/hda_codec.c:2065: BUG? (info->nid) [ 38.936946] [<f8c19bc0>] snd_hda_build_pcms+0x1e0/0x220 [snd_hda_intel] [ 38.936962] [<f8c1598c>] azx_probe+0x71c/0xa50 [snd_hda_intel] [ 38.936974] [<f8c14850>] azx_send_cmd+0x0/0x130 [snd_hda_intel] [ 38.936981] [<f8c14980>] azx_get_response+0x0/0x200 [snd_hda_intel] [ 38.936991] [<f8c146c0>] azx_power_notify+0x0/0x70 [snd_hda_intel] [ 38.937002] [<c0209a66>] pci_device_probe+0x56/0x80 [ 38.937008] [<c026110e>] driver_probe_device+0x8e/0x190 [ 38.937013] [<c026137e>] __driver_attach+0x9e/0xa0 [ 38.937017] [<c02604fb>] bus_for_each_dev+0x3b/0x60 [ 38.937022] [<c0260f86>] driver_attach+0x16/0x20 [ 38.937024] [<c02612e0>] __driver_attach+0x0/0xa0 [ 38.937026] [<c02608ca>] bus_add_driver+0x8a/0x1b0 [ 38.937031] [<c0209c13>] __pci_register_driver+0x53/0xa0 [ 38.937035] [<c014a7d1>] sys_init_module+0x151/0x1a00 [ 38.937038] [<c01fb39f>] prio_tree_insert+0x1f/0x250 [ 38.937052] [<c01041d2>] sysenter_past_esp+0x6b/0xa9 [ 38.937058] ======================= [ 38.937097] ACPI: PCI interrupt for device 0000:00:1b.0 disabled [ 38.937102] HDA Intel: probe of 0000:00:1b.0 failed with error -22 [ 38.937124] ACPI: PCI Interrupt 0000:01:00.1[B] -> GSI 17 (level, low) -> IRQ 23 [ 38.937134] PCI: Setting latency timer of device 0000:01:00.1 to 64 [ 38.952813] ALSA /home/matt/Data/alsa-driver-hg20080110/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:732: codec_mask = 0x1

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[alsa-devel] a7839e96 (PNP: increase max resources) breaks my ALSA intel8x0 sound card
by Avuton Olrich 28 Jan '08

28 Jan '08

With v2.6.24 my second ALSA sound device stopped working. After bisection it says this was the offending commit. a7839e960675b549f06209d18283d5cee2ce9261 is first bad commit commit a7839e960675b549f06209d18283d5cee2ce9261 Author: Zhao Yakui <yakui.zhao(a)intel.com> Date: Wed Nov 28 16:21:21 2007 -0800 PNP: increase the maximum number of resources On some systems the number of resources(IO,MEM) returnedy by PNP device is greater than the PNP constant, for example motherboard devices. It brings that some resources can't be reserved and resource confilicts. This will cause PCI resources are assigned wrongly in some systems, and cause hang. This is a regression since we deleted ACPI motherboard driver and use PNP system driver. [akpm(a)linux-foundation.org: fix text and coding-style a bit] Signed-off-by: Li Shaohua <shaohua.li(a)intel.com> Signed-off-by: Zhao Yakui <yakui.zhao(a)intel.com> Cc: Bjorn Helgaas <bjorn.helgaas(a)hp.com> Cc: Thomas Renninger <trenn(a)suse.de> Cc: <stable(a)kernel.org> Signed-off-by: Andrew Morton <akpm(a)linux-foundation.org> Signed-off-by: Linus Torvalds <torvalds(a)linux-foundation.org> The audio device is 00:1b.0 (see my lspci -vvv output), the other audio device works fine. http://avuton.googlepages.com/v2.6-before (dmesg revision before) http://avuton.googlepages.com/v2.6-after (dmesg broken revision) http://avuton.googlepages.com/lspci-vvv http://avuton.googlepages.com/config (from the broken revision) http://avuton.googlepages.com/iomem http://avuton.googlepages.com/ioports -- avuton -- Anyone who quotes me in their sig is an idiot. -- Rusty Russell.

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[alsa-devel] Broken seq
by Alan Horstmann 27 Jan '08

27 Jan '08

Can someone help with this... During repeated testing of MIDI, something has broken the seq system, so when loading previously working sound drivers, I get a message: /proc/asound/seq/clients File not found (or similar) and sure enough it is not there. Why has it gone and how can I get it back? Thanks Alan

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[alsa-devel] installation steps for us-122l tascam driver
by Xavier garabedian 27 Jan '08

27 Jan '08

hello i'm a linux beginner and i'm trying to make the us 122l work on ubuntu. i have seen the patches that you published, but i don't know the steps to install them and make the tascam work. is there anyone who can discribe me the process, step by step ? thank you very much in advance best regards xavier garabedian

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[alsa-devel] Intel HDA (ICH8) unsupported model?
by Philipp E. Letschert 27 Jan '08

27 Jan '08

Hello, This is for the Zepto Znote 6224W Notebook (and probably others): lspci -v 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Inventec Corporation Unknown device 0040 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0500000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) lspci -vvn 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 1170:0040 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f0500000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) Enabling sound caused some headaches. I finally ended with kernel 2.6.24 and alsa 1.0.16rc1. The card was autodetected as Realtec ALC268, but no sound output. Using option model=toshiba instead, when loading the snd-hda-intel module did work, but sound has a really poor quality, with lots of crackling noises. Enabling the samplerate plugin didnt help... Please let me know, if this card is already supported and I just have errors in my configuration, or further informations are helpful for develepment. Thanks, Phil

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[alsa-devel] Usb driver correction
by LCID Fire 27 Jan '08

27 Jan '08

Hi. I hope you guys can help me with 3 problems I have. I have a usb device which has 3 streams - which all work fine (that's not the problem yet ;)). Aside from this there is also a mixer created. Alsa cannot retrieve any information from the mixer - not the minimum, nor the maximum or current value. Basicly alsa leaves the mixer in a state where it cannot be used - which leads to a question why not omnit the mixer? What I am ending up with are these 2 volumes: [4] FU [PCM Capture Volume] ch = 1, val = 0/1/1 [9] FU [PCM Capture Volume] ch = 1, val = 0/1/1 What would I have to do to disable these? Is it enough to define a usbmix_name_map in usbmixer_maps.c with {4, NULL} and {9, NULL}? Also, the usb device has a switch - from line input to phono input. I know what control message I have to send to change the switch - but where has such an additional switch to be implemented? Is it similar to snd_audigy2nx_led_get in usbmixer.c? Would be great if someone could explain these mysteries to me ;)

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[alsa-devel] Microphone dead: Gateway MX6543, hda-intel, STAC925x
by Zoran Rilak 27 Jan '08

27 Jan '08

Hello, I've searched the archives and haven't found a solution for this particular model (STAC9250). Front mic does not work. I have tried other models, aside from m2-2, and all four possible values for position_fix as well, but to no avail. If anyone is willing to tackle this, please point me to what other information I should provide in order to get to the bottom of this. I am using Ubuntu kernel 2.6.22-14 and ALSA 1.0.15 compiled from source. Here are my /proc/asound/card0/codec#* files: --- /proc/asound/card0/codec#0 --- Codec: SigmaTel STAC9250 Address: 0 Vendor Id: 0x83847634 Subsystem Id: 0x107b0367 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Power: 0x0 Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x14 Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x07 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x06 [Audio Selector] wcaps 0x300901: Stereo Connection: 3 0x02* 0x07 0x14 Node 0x07 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect Pin Default 0x40c003f3: [N/A] SPDIF In at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x08 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x424503f2: [N/A] SPDIF Out at Ext Front Conn = Optical, Color = Unknown Pin-ctls: 0x00: Connection: 2 0x05* 0x14 Node 0x09 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=0 Amp-Out vals: [0x0f 0x0f] Connection: 1 0x0f Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x04180011: [Jack] Speaker at Ext Right Conn = DIN, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 1 0x0e Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x50a103f0: [N/A] Mic at Int N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x20: IN Connection: 1 0x0e Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x90100212: [Fixed] Speaker at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x0e [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x0d 0x0d] Connection: 1 0x06 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Connection: 5 0x0b* 0x0c 0x0d 0x0a 0x11 Node 0x10 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x0810: OUT Pin Default 0x400003f1: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x12 Node 0x11 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x9033032e: [Fixed] CD at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x12 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x06 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x14 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x09 --- end codec#0 --- --- /proc/asound/card0/codec#1 --- Codec: Motorola Si3054 Address: 1 Vendor Id: 0x10573057 Subsystem Id: 0x10001 Revision Id: 0x100100 Modem Function Group: 0x1 --- end codec#1 ---

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[alsa-devel] [PATCH] Add missing additional verbs for SPDIF with AD1984 & thinkpad
by Jerone Young 26 Jan '08

26 Jan '08

Additional verbs are missing for IEC958 code for the AD1984 in thinkpads. This is identical to the verbs in AD19884. I missed these in my patch last month. Signed-off-by: Jerone Young <jerone(a)gmail.com> diff -r dcc1b9d533d3 alsa-kernel/pci/hda/patch_analog.c --- a/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 19:49:19 2008 -0600 +++ b/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 19:51:05 2008 -0600 @@ -3188,6 +3188,9 @@ static struct hda_verb ad1984_thinkpad_i {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* enable EAPD bit */ {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + /* SPDIF output selector */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ }; /* Digial MIC ADC NID 0x05 + 0x06 */

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[alsa-devel] [PATCH] Remove psuedo master from pci/hda/analog_patch.c . It does not function well!
by Jerone Young 26 Jan '08

26 Jan '08

Hi I noticed after installing 1.0.16rc-1 today that my Thinkpad T61 with AD1984 had a Master mixer now. Problem is it's not very good in that it 1) does not follow with PCM (the real master) volume well. It causes a volume disparity and does not follow or even boost PCM signal. So it just a bad mister. 2) it does not mute the volume (since PCM is the real master mixer you have to go to PCM to mute the volume!) I have fixed a problem in Gstreamer so that Gnome (gnome volume mixer) does finally work correctly for the AD1984 which does not have a Master. When it looks for a master mixer it now will fallback to PCM as a master mixer (the code is upsteam too). You can find this here: http://bugzilla.gnome.org/show_bug.cgi?id=506928 I have a feeling this is why this patch was put in. I have fixed the problem in Gnome, KDE has allways worked. just fine. But I would honestly love it if this code was removed!!! If you need a Master can you rename PCM to master for the AD1984! I have attached a patch to remove this code .. because it is so does not function well! Signed-off-by : Jerone Young <jerone(a)gmail.com> diff -r 5dfc8324ae17 alsa-kernel/pci/hda/patch_analog.c --- a/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 00:12:14 2008 -0600 +++ b/alsa-kernel/pci/hda/patch_analog.c Fri Jan 25 01:04:13 2008 -0600 @@ -170,26 +170,6 @@ static int ad198x_build_controls(struct } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); - if (err < 0) - return err; - } - - /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { - snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, - HDA_OUTPUT, spec->vmaster_tlv); - err = snd_hda_add_vmaster(codec, "Master Playback Volume", - spec->vmaster_tlv, - (spec->slave_vols ? - spec->slave_vols : ad_slave_vols)); - if (err < 0) - return err; - } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { - err = snd_hda_add_vmaster(codec, "Master Playback Switch", - NULL, - (spec->slave_sws ? - spec->slave_sws : ad_slave_sws)); if (err < 0) return err; }

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