July 2009 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Aug '09

24 Aug '09

2009/6/24 Denis Kup <dener.kup(a)gmail.com>: > 2009/6/24 Takashi Iwai <tiwai(a)suse.de>: >> At Wed, 24 Jun 2009 16:09:58 +0400, >> Denis Kup wrote: >>> >>> Hey, when i first installed alsa to my laptop, there no sound from >>> subwoofer, jacks were map very strange (2 surrounds for ex.). I try >>> all models from the list, but all don't support my subwoofer. After a >>> weak of hard work i made a patch that enables ACER_ASPIRE_7730G model >>> for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal >>> frontal speakers and internal subwoofer.You can see acer.com for more >>> info about hardware and jacks. >>> >>> It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: >> >> Can you try Emilio's patch recently posted (and pass >> model=acer-aspire-6530g) with the very latest alsa-driver-snapshot? >> This essentially includes your changes. It's in: >> From: Emilio López <buhitoescolar(a)gmail.com> >> Subject: Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G >> Date: Tue, 23 Jun 2009 22:32:45 -0300 >> Message-ID: <d54969c50906231832w48ce7315uc5cf8b3f81534c76(a)mail.gmail.com> >> >>> After changing file i install driver, and it automatically detect its >>> model (maybe there are old settings from previous install, so i dont >>> 100% sure that it was detected). With this settings subwoofer works >>> perfect, internal stereo and subwoofer mute when headphone are plugged >>> all works, except 1 thing: subwoofer uses center channel instead LFE. >>> To fix it i use such asound.conf file: >> ... >>> Is there any chance to swap Center and LFE in patch_realtek.c? >> >> Unlikely as ALC888* doesn't support the channel-swap feature (although >> some other ALC codecs do). >> >> >> thanks, >> >> Takashi >> > > You are talking about that > http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html > ?? Its right link to drivers http://www.alsa-project.org/snapshot/ ?? > And there is easier way to apply patches than manually adding strings > to files? > I manage with installing. I use alsa-driver-snapshot.tar.gz http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/ and this Amilo changes http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html . But this patch has smaller functionality against i have. There is advantages and disadvantages of both patches: My patch: + 1) when black jack plugged, internal speakers and subwoofer mute + 2) there is 3 maps for jacks: (black/pink/blue) 2ch: front/ext mic/ (something i dont know, it should be line in), 4ch: front/ ext mic/ side 6ch: front/ CLFE / side; (its changes in channel mode option in mixer) + 3) internal mic works + 4) there are separate boost mic channels for internal and external mics - 5) internal subwoofer controls with Center channel in mixer ? 6) i dont know how line in works (i will check it when reinstall to my patch) ? 7) i dont know how spdif works, because i have not any spdif speakers in house Amilo patch: - 1) when black jack pluged, only internal speaker mutes, subwoofer continue to work (it has no meaning when i listen to music in headphones) - 2) there is only 1 map for jacks: front/ext mic/ (something i dont know, it should be line in) - 3) there no side and center channels - 3) internal mic dont work - 4) there no line in in input source menu in mixer + 5) Subwoofer controls with LFE channel in mixer ? 7) i dont know how spdif works, because i have not any spdif speakers in house

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[alsa-devel] Support for ALC662 on FUJITSU SIEMENS AMILO Li3710
by Mads Kiilerich 20 Aug '09

20 Aug '09

On a friends FUJITSU SIEMENS AMILO Li3710 10601011427 sound didn't work with kernel-PAE-2.6.29.6-213.fc11.i686 Out of the box we get no sound and HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 hda_codec: Unknown model for ALC662, trying auto-probe from BIOS... ALSA sound/pci/hda/hda_codec.c:3507: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3511: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3515: hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3516: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3524: inputs: mic=0x18, fmic=0x19, line=0x1a, fline=0x0, cd=0x0, aux=0x0 But with /etc/modprobe.d/local.conf (from http://ubuntuforums.org/archive/index.php/t-940689.html) options snd-hda-intel model=3stack-dig options snd-hda-intel single_cmd=1 options snd-hda-intel enable_msi=1 we get sound and HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: irq 32 for MSI/MSI-X HDA Intel 0000:00:1b.0: setting latency timer to 64 ALSA sound/pci/hda/hda_codec.c:2698: hda_codec: model '3stack-dig' is selected I would like to have this fixed so Linux sound works on my friends machine without manual configuration. After having read some documentation I assume that single_cmd and enable_msi perhaps shouldn't be needed. I will test that next time I get access to that machine. HAL reports the sound card as /org/freedesktop/Hal/devices/pci_8086_293e, so should this patch be sufficient? diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e1563f4..0385353 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -16699,6 +16699,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", ALC663_ASUS_H13), + SND_PCI_QUIRK(0x8086, 0x293e, "FUJITSU SIEMENS AMILO Li3710", + ALC662_3ST_2ch_DIG), {} }; Untested and hard to test because I don't have direct access to the machine. This issue was also reported on https://bugzilla.redhat.com/show_bug.cgi?id=487889 Please CC me. /Mads

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[alsa-devel] [PATCH] Allow passing platform_data to devices attached to AC97 bus
by Marek Vasut 14 Aug '09

14 Aug '09

Hi, I managed to put together this patch with huge help from Mark. It allows passing platform_data to devices on ac97 bus and implements this for pxa2xx- ac97. Please consider applying.

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[alsa-devel] [PATCH 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 13 Aug '09

13 Aug '09

Hellos guys, This series provides a way to allow DMA go to idle while performing playback/capture in OMAP3 devices using ASoC driver. The idea behind it is to do not use DMA element mode, but frame mode. In addition to that, it also configures McBSP threshold accordingly to alsa period. Also, McBSP wakeup signals are configured to generate wakeups for threshold events. Besides that, this series also include a way to allow choosing between element mode or the frame mode added here. Patches 1-15 are McBSP changes and, for this reason, are intended to be include in linux-omap tree. Patches 16-20 are OMAP ASoC driver changes, so, they are for alsa-devel. BR, Eduardo Valentin (11): OMAP: McBSP: Change wakeup signals OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Use McBSP threshold again ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Disable all wakeups upon exit OMAP: McBSP: Let element DMA mode hit retention also ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 54 +++++ arch/arm/plat-omap/mcbsp.c | 343 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 16 ++- sound/soc/omap/omap-pcm.c | 57 +++++- 5 files changed, 468 insertions(+), 7 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] load_for_all_cards bug
by Kevin Goodsell 10 Aug '09

10 Aug '09

(I originally tried to send this via the moderator without subscribing to the list, but it never went through and I never received a moderator response. Apparently this list is set up as moderated, but nobody is actually moderating it.) Summary: load_for_all_cards fails with existing configuration elements In snd_config_hook_load_for_all_cards, the first call to snd_config_search attempts to locate an existing configuration node with the name of the driver. Typically none is found, and everything is good. However, if such a node is located, the next line assumes it is a leaf node with type 'string' and calls snd_config_get_string to fetch the string value. If this fails, the entire hook is abandoned. Because of this, setting something like the following in asoundrc: cards.<driver name>.foo 0 is sufficient to disable the entire card-specific configuration. As a concrete example, I have a HDA-Intel sound card. dmix.conf includes a way to set period_size, period_time, and periods by using configuration elements of the form cards.<driver name>.pcm.dmix.<var>. In ~/.asoundrc I add cards.HDA-Intel.pcm.dmix.period_size 1024 This will cause HDA-Intel.conf to fail to load, and the pcm defined in default.conf will fail to find the device-specific pcm cards.HDA-Intel.pcm.default, and fall back on the default pcm using plughw. By attempting to configure dmix, I have disabled it. Fixing this may just be a matter of changing the line in snd_config_hook_load_for_all_cards from: if (snd_config_search(root, fdriver, &n) >= 0) { to: if (snd_config_search(root, fdriver, &n) >= 0 && n->type != SND_CONFIG_TYPE_COMPOUND) { I have not tested this. -Kevin

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[alsa-devel] [BUG] NULL pointer dereference in patch_sigmatel.c
by Ozan Çağlayan 10 Aug '09

10 Aug '09

Hi, One of our users is having a NULL ptr dereference upon loading the snd_hda_intel module with 20090624's snapshot. There's only one commit after that date in patch_sigmatel.c so I didn't tell him to try with the latest snapshot but if you think that the bug may be related to another part of the ALSA codebase, I can make him try the latest snapshot. I'm attaching the alsa-info output on a working system with 1.0.18a. The kernel version in subject is 2.6.25.20. Thanks, Ozan ------------------------ First the BUG's trace: ------------------------ BUG: unable to handle kernel NULL pointer dereference at 00000074 IP: [<f8cdb83a>] :snd_hda_codec_idt:stac92xx_add_jack+0x84/0x9e *pde = 00000000 Oops: 0002 [#1] SMP Modules linked in: snd_hda_codec_idt snd_hda_intel(+) snd_hda_codec aes_i586 aes_generic ipv6 af_packet bridge bnep rfcomm l2cap microcode acpi_cpufreq cpufreq_powersave cpufreq_userspace cpufreq_conservative ndiswrapper vboxdrv pcmcia snd_hwdep rtc_cmos rtc_core yenta_socket tpm_infineon snd_seq_dummy rtc_lib tpm snd_seq_oss rsrc_nonstatic tpm_bios tifm_7xx1 iTCO_wdt snd_seq_midi_event tifm_core i2c_i801 snd_seq iTCO_vendor_support pcmcia_core r5u870 arc4 joydev snd_seq_device snd_pcm_oss nvidia(P) battery sg usbcam snd_mixer_oss ecb ac videobuf_dma_sg videobuf_core sony_laptop snd_pcm uvcvideo iwl4965 video compat_ioctl32 snd_timer output hci_usb iwlcore videodev thermal bluetooth snd processor v4l1_compat rfkill button led_class soundcore firmware_class snd_page_alloc mac80211 intel_agp i2c_core sky2 cfg80211 agpgart ext3 jbd mbcache sr_mod cdrom sd_mod ata_piix uhci_hcd pata_acpi ehci_hcd usbcore ohci1394 ieee1394 ata_generic libata scsi_mod dock Pid: 1977, comm: modprobe Tainted: P (2.6.25.20-114 #1) EIP: 0060:[<f8cdb83a>] EFLAGS: 00210206 CPU: 0 EIP is at stac92xx_add_jack+0x84/0x9e [snd_hda_codec_idt] EAX: 00000000 EBX: f8c7973e ECX: 00000004 EDX: f8cdeb1a ESI: 03211020 EDI: f675c600 EBP: f68d7d20 ESP: f68d7cd4 DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Process modprobe (pid: 1977, ti=f68d6000 task=f69ece60 task.ti=f68d6000) Stack: f68d7cf4 f8cdeb0a f8c796d5 f8c7973e f8c79793 00000001 000a69e0 f8c79793 4f205048 61207475 78452074 654c2074 4a207466 006b6361 f6898800 f6898800 00000000 f6898800 f6898800 f68d7d4c f8cdbac0 f65ef000 f6970000 00000001 Call Trace: [<f8cdbac0>] ? stac92xx_build_controls+0x26c/0x2d0 [snd_hda_codec_idt] [<f8c73aa9>] ? snd_hda_codec_build_controls+0x31/0x3d [snd_hda_codec] [<f8c750d3>] ? snd_hda_build_controls+0x18/0x67 [snd_hda_codec] [<f8c69138>] ? azx_probe+0x79e/0x840 [snd_hda_intel] [<f8c68857>] ? azx_send_cmd+0x0/0x101 [snd_hda_intel] [<f8c686a6>] ? azx_get_response+0x0/0x1b1 [snd_hda_intel] [<f8c67ece>] ? azx_attach_pcm_stream+0x0/0x15c [snd_hda_intel] [<f8c67b84>] ? azx_bus_reset+0x0/0x56 [snd_hda_intel] [<f8c67a40>] ? azx_power_notify+0x0/0x57 [snd_hda_intel] [<c01e7a37>] ? pci_device_probe+0x39/0x59 [<c024395f>] ? driver_probe_device+0xa0/0x136 [<c0243a50>] ? __driver_attach+0x5b/0x91 [<c024333c>] ? bus_for_each_dev+0x3b/0x63 [<c0243804>] ? driver_attach+0x14/0x16 [<c02439f5>] ? __driver_attach+0x0/0x91 [<c0242d3a>] ? bus_add_driver+0x9d/0x1ba [<c0243bc4>] ? driver_register+0x47/0xa7 [<c0168681>] ? __vunmap+0x93/0x9b [<c01e7bec>] ? __pci_register_driver+0x35/0x61 [<f8afd017>] ? alsa_card_azx_init+0x17/0x19 [snd_hda_intel] [<c0141f9c>] ? sys_init_module+0x18ad/0x19ca [<c0109c77>] ? do_syscall_trace+0x138/0x17f [<c0104a2e>] ? syscall_call+0x7/0xb ======================= Code: f9 ff 89 45 d0 89 f0 e8 29 68 f9 ff 89 c3 89 f0 e8 32 68 f9 ff ff 75 d0 53 50 68 0a eb cd f8 8d 45 d4 50 e8 45 25 50 c7 8b 47 08 <89> 78 74 8b 47 08 83 c4 14 c7 40 78 7f a2 cd f8 31 c0 8d 65 f4 EIP: [<f8cdb83a>] stac92xx_add_jack+0x84/0x9e [snd_hda_codec_idt] SS:ESP 0068:f68d7cd4 ---[ end trace 4a628a5b4fc302e7 ]--- alsa-info output: ------------------- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.52 !!################################ !!Script ran on: Sun Feb 8 19:15:27 EET 2009 !!Linux Distribution !!------------------ Pardus 2008.2 Canis aureus !!Kernel Information !!------------------ Kernel release: 2.6.25.20-114 Operating System: GNU/Linux Architecture: i686 Processor: Intel(R) Core(TM)2 Duo CPU T8100 @ 2.10GHz SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.18a Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc300000 irq 21 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) 09:04.2 Mass storage controller: Texas Instruments 5-in-1 Multimedia Card Reader (SD/MMC/MS/MS PRO/xD) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 104d:9008 !!Modprobe options (Sound related) !!-------------------------------- snd-hda-intel: model=vaio !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : vaio,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: SigmaTel STAC9872AK Address: 0 Vendor Id: 0x83847662 Subsystem Id: 0x104d1e00 Revision Id: 0x100201 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: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=5, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x06 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x07 Processing caps: benign=0, ncoeff=0 Node 0x07 [Audio Selector] wcaps 0x300903: Stereo Amp-In R/L Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x0e Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x09 Processing caps: benign=0, ncoeff=0 Node 0x09 [Audio Selector] wcaps 0x300903: Stereo Amp-In R/L Amp-In caps: N/A Amp-In vals: [0x0d 0x0d] Connection: 1 0x15 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x03211020: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x80: HP VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 1 0x02 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x03a15030: [Jack] Mic at Ext Left Conn = 1/8, Color = Red DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x05 Node 0x10 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x3e0]: 44100 48000 88200 96000 176400 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x11 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 2 0x10* 0x09 Node 0x12 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x13 Node 0x13 [Pin Complex] wcaps 0x440381: Stereo Digital Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Delay: 4 samples Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x90a7013e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0xe Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=1 Amp-Out vals: [0x00 0x00] Connection: 4 0x0a 0x0d 0x14* 0x02 Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x17 [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=0, val=127 Connection: 4 0x02* 0x03 0x04 0x05 Node 0x18 [Audio Output] wcaps 0x40201: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 Delay: 4 samples Codec: Conexant ID 2c06 Address: 1 Vendor Id: 0x14f12c06 Subsystem Id: 0x104d1700 Revision Id: 0x100000 Modem Function Group: 0x2 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Şub 8 2009 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Şub 8 2009 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 Şub 8 2009 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 24 Şub 8 19:14 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Şub 8 19:15 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Şub 8 2009 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Şub 8 2009 /dev/snd/timer !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # PulseAudio plugin configuration # Let's create a virtual device "pulse" for mixer and PCM pcm.pulse { type pulse hint { description "PulseAudio Sound Server" } } ctl.pulse { type pulse hint { description "PulseAudio Sound Server" } } # Let's make it the default! pcm.!default { type pulse hint { description "Default" } } ctl.!default { type pulse hint { description "Default" } } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xfc300000 irq 21' Mixer name : 'SigmaTel STAC9872AK' Components : 'HDA:83847662,104d1e00,00100201 HDA:14f12c06,104d1700,00100000' Controls : 10 Simple ctrls : 7 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 127 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 127 [100%] [0.00dB] [on] Front Right: Playback 127 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 255 Mono: Capture [off] Front Left: Playback 0 [0%] [-51.00dB] Front Right: Playback 0 [0%] [-51.00dB] Simple mixer control 'Mic Jack',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 13 [87%] [19.50dB] [on] Front Right: Capture 13 [87%] [19.50dB] [on] Simple mixer control 'Internal Mic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 127 [100%] [0.00dB] [on] Front Right: Playback 127 [100%] [0.00dB] [on] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Headphone Playback Volume' value.0 127 value.1 127 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Speaker Playback Volume' value.0 127 value.1 127 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' comment.dbmin 0 comment.dbmax 2250 iface MIXER name 'Capture Volume' value.0 13 value.1 13 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.7 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'Mic Jack' comment.item.1 'Internal Mic' comment.item.2 PCM iface MIXER name 'Capture Source' value 'Internal Mic' } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 127 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.10 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module aes_i586 aes_generic ipv6 bridge bnep af_packet rfcomm l2cap microcode acpi_cpufreq cpufreq_powersave cpufreq_userspace ndiswrapper r5u870 usbcam videobuf_dma_sg snd_hda_intel snd_seq_dummy snd_seq_oss snd_seq_midi_event videobuf_core nvidia snd_seq uvcvideo snd_seq_device snd_pcm_oss arc4 compat_ioctl32 ecb snd_mixer_oss intel_agp snd_pcm iwl4965 snd_timer snd_page_alloc snd_hwdep snd pcmcia hci_usb rtc_cmos bluetooth yenta_socket rsrc_nonstatic tpm_infineon pcmcia_core tifm_7xx1 sky2 battery videodev iTCO_wdt agpgart ac thermal processor iwlcore rfkill led_class firmware_class i2c_i801 mac80211 video output tpm tpm_bios rtc_core rtc_lib joydev iTCO_vendor_support tifm_core sony_laptop v4l1_compat i2c_core sg button soundcore cfg80211 ext3 jbd mbcache sr_mod cdrom sd_mod ata_piix ohci1394 ieee1394 uhci_hcd pata_acpi ata_generic libata scsi_mod dock ehci_hcd usbcore

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Frashman 09 Aug '09

09 Aug '09

Hi Takashi, first of all, thank you for developing this driver. I tested it today. It works, so far. But two annoying things. As you mentioned, I got problems, if the onboard sound modules are loaded. Then i get no sound out auf my xfi. If I disable the loading of snd-hda-intel modules while booting, then it works. The second problem is, that SIDE and SURROUND connectors are interchanged. I'm absolutly sure, it works fine under windows and the connector colors are also correct. If I run speaker-test -c8 the SIDE Speakers sounds come out of my REAR (SURROUND) Speakers, while nothing is coming out of my rear speakers, if i'm using speaker-test -c6 Here lspci: $ lspci 00:00.0 Host bridge: ATI Technologies Inc RD780 Northbridge only dual slot PCI-e_GFX and HT1 K8 part 00:02.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (external gfx0 port A) 00:04.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (PCI express gpp port A) 00:0a.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (PCI express gpp port F) 00:12.0 SATA controller: ATI Technologies Inc SB600 Non-Raid-5 SATA 00:13.0 USB Controller: ATI Technologies Inc SB600 USB (OHCI0) 00:13.1 USB Controller: ATI Technologies Inc SB600 USB (OHCI1) 00:13.2 USB Controller: ATI Technologies Inc SB600 USB (OHCI2) 00:13.3 USB Controller: ATI Technologies Inc SB600 USB (OHCI3) 00:13.4 USB Controller: ATI Technologies Inc SB600 USB (OHCI4) 00:13.5 USB Controller: ATI Technologies Inc SB600 USB Controller (EHCI) 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 14) 00:14.1 IDE interface: ATI Technologies Inc SB600 IDE 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 00:14.3 ISA bridge: ATI Technologies Inc SB600 PCI to LPC Bridge 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:00.0 VGA compatible controller: ATI Technologies Inc RV770 [Radeon HD 4850] 01:00.1 Audio device: ATI Technologies Inc HD48x0 audio 02:00.0 Audio device: Creative Labs X-Fi Titanium series [EMU20k2] (rev 03) 03:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 01) 04:06.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 04:06.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 04:0e.0 FireWire (IEEE 1394): Texas Instruments TSB43AB23 IEEE-1394a-2000 Controller (PHY/Link) Thank you so far, Frashman

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[alsa-devel] [PATCH] ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6
by Candelaria Villareal, Jorge 07 Aug '09

07 Aug '09

Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6 line, controlled by register INTBR_PMBR1. Machine driver takes care of enabling gpio line through i2c and codec driver manipulates the line during headset ramp up/down sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index c51594d..f7e5b74 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -24,6 +24,7 @@ #include <linux/clk.h> #include <linux/platform_device.h> +#include <linux/i2c/twl4030.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -39,6 +40,9 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" +#define TWL4030_INTBR_PMBR1 0x0D +#define EXTMUTE(value) (value << 2) + static struct snd_soc_card snd_soc_sdp3430; static int sdp3430_hw_params(struct snd_pcm_substream *substream, @@ -280,6 +284,7 @@ static struct snd_soc_card snd_soc_sdp3430 = { static struct twl4030_setup_data twl4030_setup = { .ramp_delay_value = 3, .sysclk = 26000, + .hs_extmute = 1, }; /* Audio subsystem */ @@ -312,6 +317,10 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ + /* Set TWL4030 GPIO6 as EXTMUTE signal */ + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), + TWL4030_MODULE_INTBR); + ret = platform_device_add(sdp3430_snd_device); if (ret) goto err1; -- 1.6.0.4

3 6

[alsa-devel] [PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 07 Aug '09

07 Aug '09

We've encountered strange behavior in the alsamixer settings using the wm9712 codec. If we unmute the headphone output and then unmute the PCM output, the headphone output gets reset to mute in the hardware register. At this point the hardware register does not match the value in the register cache. I've spent some time debugging this, and the headphone setting is set outside of any code path that would call the ac97_write() routine. As best as I can tell, there is something strange going on in hardware. I've provided this patch that works around the problem. Have any of you seen this before? Is this patch the right approach? - John The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task))

5 14

[alsa-devel] [PATCH 1/3] ASoC: change set_tdm_slot api to allow slot_width override.
by Daniel Ribeiro 06 Aug '09

06 Aug '09

Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc-dai.h | 5 +++-- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++++++++------- sound/soc/soc-core.c | 9 ++++++--- 5 files changed, 33 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..6692410 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e..c47d573 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1206,8 +1206,9 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +/* FIXME: Needs to handle slot_width */ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1219,7 +1220,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cd..34437d8 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e22c5ce..5b9ed64 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..c72a8e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } -- tg: (c787980..) asoc/set_tdm_slot (depends on: asoc/remove-code-duplication) -- Daniel Ribeiro

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