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

[alsa-devel] get two mono channel from stereo line
by arif setiawan 27 Jul '09

27 Jul '09

Thank you. It works as we expected, however we found that we can't run two lines (left-capture - left-playback) and (right-capture - right-playback) simulatenously. The error message is : ALSA lib pcm_dshare.c:706:(snd_pcm_dshare_open) unable to open slave cannot open output audio device right: Device or resource busy Is there any workaround for this?

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Re: [alsa-devel] Getting around initial sample loss with intel HDMI
by Takashi Iwai 27 Jul '09

27 Jul '09

At Sun, 26 Jul 2009 23:21:50 -0700, Shane W wrote: > > On Mon, Jul 27, 2009 at 08:07:40AM +0200, Takashi Iwai wrote: > > At Sun, 26 Jul 2009 10:32:10 -0700, > > Shane W wrote: > > > I did some digging in the drivers to try and find the > > > source of this problem. Note that I am really not familiar > > > with this stuff. It appears to be here: > > > hda_codec.c > > > void snd_hda_codec_setup_stream(...) > > > ... > > > snd_hda_codec_write(codec, nid, 0, > > > AC_VERB_SET_CHANNEL_STREAMID, > > > > > > Introducing a delay after this point fixes the issue but > > > it's around a full second so that's obviously not a > > > solution. Could we not cache the current format and only do > > > this when the format changes rather than on every device > > > open? > > > > The format itself is kept. But the stream tag and channel assignment > > are assigned dynamically at each open. Thus, this call has to be > > reset at each open, at least. > > > > > For testing, I tried doing this though and disabling > > > the set_streamid, 0 call in snd_codec_cleanup() but it > > > still missed those first samples. Is there somewhere else I > > > need to be looking? > > > > Does it happen without dmix, too? > > Yeah, with and without dmix it still misses the first bit. OK. I just thought of a possibility of dmix bug. Good to know that it's not :) > The only way it doesn't is if I have another program open > the device and sleep, then have the samples play. Actually, > an open isn't sufficient, it has to go and set the playback > parameters as well. It's logical since the stream and format ids are assigned dynamically first after the stream is set up. One another problem is that we currently call azx_stream_reset() in each prepare callback in hda_intel.c. This was newly added because keeping the format id without resetting caused some other serious errors. Maybe this can be optimized for specific hardwares, e.g. skipping the unneeded reset for HDMI/SPDIF. Takashi

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Re: [alsa-devel] Getting around initial sample loss with intel HDMI
by Takashi Iwai 27 Jul '09

27 Jul '09

At Sun, 26 Jul 2009 10:32:10 -0700, Shane W wrote: > > On Sun, Jul 26, 2009 at 10:52:01AM +0200, Takashi Iwai wrote: > > > It's a known issue that Intel HDMI audio loses the first > > > half second or so of audio when opening the device. Using > > > the dmix interface, I've been able to get around that by > > > writing a quick app to hold the device open so the HDMI > > > stays live. However, with the recent asound.conf posted > > > here which uses the route plugin to fix the multichannel > > > order, I can't do that as dmix isn't in the picture. I'd > > > rather not use dmix anyway since it can bugger with the > > > sample rate so I am wondering if there's another way to get > > > around this. > > > > It's possible to combine with dmix. > > You set up dmix with 8 channels, put route plugin over it. > > This has a few problems. > > It doesn't seem to handle varying sample rates. IE if I > play a 96khz stream at 6 channels, sound is terrible. This is rather a problem of Nvidia HDMI codec driver... > Also, > streaming DTS or AC3 no longer works since the streams are > being upmixed to 6ch. This has to be sent anyway exclusively without dmix, so you'll need another PCM definition. > I did some digging in the drivers to try and find the > source of this problem. Note that I am really not familiar > with this stuff. It appears to be here: > hda_codec.c > void snd_hda_codec_setup_stream(...) > ... > snd_hda_codec_write(codec, nid, 0, > AC_VERB_SET_CHANNEL_STREAMID, > > Introducing a delay after this point fixes the issue but > it's around a full second so that's obviously not a > solution. Could we not cache the current format and only do > this when the format changes rather than on every device > open? The format itself is kept. But the stream tag and channel assignment are assigned dynamically at each open. Thus, this call has to be reset at each open, at least. > For testing, I tried doing this though and disabling > the set_streamid, 0 call in snd_codec_cleanup() but it > still missed those first samples. Is there somewhere else I > need to be looking? Does it happen without dmix, too? The reset part can be a bit more optimized. But, this is really a sensitive part, which can easily break the I/O status without knowing... Takashi

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Re: [alsa-devel] Getting around initial sample loss with intel HDMI
by Takashi Iwai 26 Jul '09

26 Jul '09

At Sat, 25 Jul 2009 13:35:18 -0700, Shane W wrote: > > Hi, > > It's a known issue that Intel HDMI audio loses the first > half second or so of audio when opening the device. Using > the dmix interface, I've been able to get around that by > writing a quick app to hold the device open so the HDMI > stays live. However, with the recent asound.conf posted > here which uses the route plugin to fix the multichannel > order, I can't do that as dmix isn't in the picture. I'd > rather not use dmix anyway since it can bugger with the > sample rate so I am wondering if there's another way to get > around this. It's possible to combine with dmix. You set up dmix with 8 channels, put route plugin over it. Takashi

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[alsa-devel] [snd_via82xx] EAGAIN on snd_pcm_prepare after EPIPE on mmap_commit
by rom1dep 25 Jul '09

25 Jul '09

Hello ! I am encountering some issues with pulseaudio : after some time and for no evident reason PA stops routing the sound. That's what I get with $pulseaudio -vvv : http://pastebin.mandriva.com/13648 (got from a 2.6.29.6_mdv kernel) running $pacmd list-sinks when stuck gives : http://pastebin.mandriva.com/13649 Our pulseaudio integrator Colin Guthrie concluded : "OK, so what is happening is that some alsa level problem is occurring (due to mmap). Pulse tries to recover from this but cannot. The alsa sink (and the whole card) is subsequently unloaded due to this fatal error." And now.. a quick paste from irc : "[00:44:20] <coling> Jul 22 14:04:29 <mezcalero> coling: driver issue i am pretty sure [00:44:20] <coling> Jul 22 14:04:33 <mezcalero> i saw that before once [00:44:20] <coling> Jul 22 14:04:55 <mezcalero> coling: i.e. EAGAIN on snd_pcm_prepare after EPIPE on mmap_commit [00:44:23] <coling> Jul 22 14:05:08 <mezcalero> coling: needs to be brought to attention of alsa folks i guess [00:44:30] <coling> That's what I was told." I'm now on 2.6.30 and I can reproduce that bug, then I'm bringing that to you ! :) Thanks !

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[alsa-devel] Fw: ALSA Volume control with Creative X
by yes work 25 Jul '09

25 Jul '09

----- Forwarded Message ---- From: yes work <yeswork(a)yahoo.com> To: clemens(a)ladisch.de Sent: Wednesday, July 1, 2009 13:03:22 Subject: ALSA Volume control with Creative X Hi, I read in one of your posts about an interest info to improve ALSA. I would like to provide some information that might help improve/augment ALSA for me and others using a creative external usb sound "card." I am running Fedora 11, command line (no windowing environment) and I have a Creative Xmod external USB sound card http://us.creative.com/products/product.asp?category=209&subcategory=668&pr… This Creative device outputs sound OK through ALSA, but the Volume control/Mute knob DO NOT CHANGE the audio. Creative does not provide Linux drivers (the knob works to control volume and mute without installing any drivers on a MS Windows setup). In Fedora, turning the volume knob causes "escape characters" to be displayed on the command line. I have included a copy of an cropped lsusb -v log, below. I hope this might be a step towards getting the volume/mute knob to work with ALSA/Fedora. Let me know if I can be of more help. Thanks, Keith [vortexbox.localdomain ~]# lsusb -v Bus 002 Device 002: ID 041e:30d0 Creative Technology, Ltd Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 64 idVendor 0x041e Creative Technology, Ltd idProduct 0x30d0 bcdDevice 1.00 iManufacturer 1 Creative Technology Ltd iProduct 2 Creative Xmod iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 315 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 98mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 159 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 6 wChannelConfig 0x003f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 12 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 17 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 11 bNrInPins 2 baSourceID( 0) 1 baSourceID( 1) 13 bNrChannels 6 wChannelConfig 0x003f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) iChannelNames 0 bmControls 0x00 bmControls 0x00 iMixer 0 junk at descriptor end: 00 00 00 AudioControl Interface Descriptor: bLength 18 bDescriptorType 36 bDescriptorSubtype 7 (PROCESSING_UNIT) bUnitID 10 wProcessType 1 bNrPins 1 baSourceID( 0) 11 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 bControlSize 1 bmControls( 0) 0x01 Enable Processing iProcessing 0 Process-Specific 01 03 00 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 5 bSourceID 10 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 6 bSourceID 4 bControlSize 1 bmaControls( 0) 0x00 bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 7 bControlSize 1 bmaControls( 0) 0x40 Automatic Gain bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 13 bSourceID 12 bControlSize 1 bmaControls( 0) 0x01 Mute bmaControls( 1) 0x02 Volume bmaControls( 2) 0x02 Volume iFeature 0 AudioControl Interface Descriptor: bLength 8 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 8 bNrInPins 2 baSource( 0) 6 baSource( 1) 9 iSelector 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 8 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 6 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x021c 1x 540 bytes bInterval 1 bRefresh 0 bSynchAddress 129 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 5 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 0 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x00b4 1x 180 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.11 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 45 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0001 1x 1 bytes bInterval 10 Device Status: 0x0000 (Bus Powered) [vortexbox.localdomain ~]# New Email names for you! Get the Email name you've always wanted on the new @ymail and @rocketmail. Hurry before someone else does! http://mail.promotions.yahoo.com/newdomains/aa/ Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/

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[alsa-devel] Help with multichannel High Speed USB audio device
by Laurent . 25 Jul '09

25 Jul '09

Dear list, I made an embedded device that has a CODEC with 6 inputs connected to an ARM9 micro-controller that has an integrated high-speed USB controller. Now, I need to write the USB code on that micro-controller so that i can capture those channels in real-time at 24bits / 48 KHz with low latency (ie. sub 10ms), on a computer. I only need my device to work with Linux, and if possible, I would like to avoid writing a device driver on the Linux-side. Is there already such a USB audio driver available that I could nearly use "as is" and to which i would conform ? Would you have some general guideline on how to write the high-speed USB device code ? Would you have any comment on the following : For simplicity, my device is going to be the "clock master" of the system since it is already has the exact clock that drives the CODEC. I will maintain a circular buffer of samples (well, frames of samples) coming from the CODEC. Each time a new frame arrives, i put the samples in the buffer and increase a counter so i know how many frames of samples I have in the buffer, ready to be sent. Each USB micro-frame (125ms/8000Hz), I send all the bytes corresponding to the frames that I have in the circular buffer. In average, 6*3*48000/8000 = 108, but I presume that could jitter slightly, which should not be a problem at all. Thanks in advance, Laurent _________________________________________________________________ Windows Live™ Hotmail®: Celebrate the moment with your favorite sports pics. Check it out. http://www.windowslive.com/Online/Hotmail/Campaign/QuickAdd?ocid=TXT_TAGLM_…

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[alsa-devel] ich10 usb_host vs snd_usb_audio Hercules - irq broadcast.
by Alexey Fisher 24 Jul '09

24 Jul '09

Hallo, i found some time to investigate the issue with my "Hercules USB 5.1 Cannel Audio Adapter". The issue is pretty tricky and depend to host hardware, in my situation this is Intel DG45ID board. Symptoms are: if i use this sound adapter after some time i'll get: irq 16: nobody cared ... Disabling IRQ #16. After closer look i found, it depend on usb port. And if i use one "wrong" usb port on IRQ 18 it will broadcast interrupts to two IRQs 18 and 16, until IRQ 16 will die. The problem is... this happens only with this sound card. So this is not easy to locate bug. And it's working fine on Win Vista and on Mac OS on same hardware.. so it should be software issue. Here is a copy of my post on http://bugzilla.kernel.org/show_bug.cgi?id=13816 If use any usb connector located on irq 18 for usb soundcard it will cause "irq 16: nobody cared", so every thing connected (usb mouse, keyboard ) to irq 16 will be broken. If i connect devices with less traffic (like mouse or keyboard) to irq 18, and usb soundcard to irq 16 or 21 ... every thing will be fine. irqpoll do not do any difference. / irq16|uhci-3 (mouse, keyboard) irq18|ehci-1- irq21|uhci-4 (sound work fine normal) \ irq18|uhci-5 (sound will brake uhci-3) / irq23|uhci-6 (not used) irq23|ehci-2- irq19|uhci-7 (not used) \ irq18|uhci-8 (not used) ========================================================================= [ 315.056793] irq 16: nobody cared (try booting with the "irqpoll" option) [ 315.056799] Pid: 0, comm: swapper Not tainted 2.6.31-rc3-00216-gea11951-dirty #98 [ 315.056802] Call Trace: [ 315.056804] <IRQ> [<ffffffff8109d666>] __report_bad_irq+0x26/0xa0 [ 315.056814] [<ffffffff8109d86c>] note_interrupt+0x18c/0x1d0 [ 315.056819] [<ffffffff8106a373>] ? sched_clock_tick+0x73/0xb0 [ 315.056823] [<ffffffff8109deb5>] handle_fasteoi_irq+0xb5/0xe0 [ 315.056827] [<ffffffff8100dffd>] handle_irq+0x1d/0x30 [ 315.056830] [<ffffffff8100d527>] do_IRQ+0x67/0xe0 [ 315.056834] [<ffffffff8100c013>] ret_from_intr+0x0/0xa [ 315.056837] <EOI> [<ffffffff812822e1>] ? acpi_idle_enter_bm+0x256/0x288 [ 315.056844] [<ffffffff812822de>] ? acpi_idle_enter_bm+0x253/0x288 [ 315.056849] [<ffffffff813a480b>] ? cpuidle_idle_call+0x9b/0xf0 [ 315.056853] [<ffffffff8100a59c>] ? cpu_idle+0xac/0x100 [ 315.056858] [<ffffffff81496825>] ? rest_init+0x65/0x70 [ 315.056862] [<ffffffff8172ed35>] ? start_kernel+0x3ba/0x3c5 [ 315.056866] [<ffffffff8172e315>] ? x86_64_start_reservations+0x125/0x129 [ 315.056870] [<ffffffff8172e3fd>] ? x86_64_start_kernel+0xe4/0xeb [ 315.056872] handlers: [ 315.056874] [<ffffffff8134bdb0>] (usb_hcd_irq+0x0/0x70) [ 315.056878] Disabling IRQ #16 ================================================================== I did smole script to test haw interrupts are used: $for i in $(seq 1 1000); do cat /proc/interrupts >> test; sleep 10s; done $grep usb[3,4,5] test there is no connections on usb3. on usb4 are keyboard an mouse. on usb5 is the soundcard. ====================================================================== -------after boot. there is too match interrupts on 16, -------there is nothing connected. 16: 4276 4520 IO-APIC-fasteoi uhci_hcd:usb3 18: 2732 2598 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 856 816 IO-APIC-fasteoi uhci_hcd:usb4 -------here i begin to play to usb sound card on usb5. 16: 12162 10987 IO-APIC-fasteoi uhci_hcd:usb3 18: 8070 7134 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 887 875 IO-APIC-fasteoi uhci_hcd:usb4 16: 20097 17784 IO-APIC-fasteoi uhci_hcd:usb3 18: 13539 11674 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1003 958 IO-APIC-fasteoi uhci_hcd:usb4 ------- count of interrupts on irq 16 is almost double of irq18 16: 28083 24881 IO-APIC-fasteoi uhci_hcd:usb3 18: 19091 16130 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1005 968 IO-APIC-fasteoi uhci_hcd:usb4 ...... ...... 16: 94758 81082 IO-APIC-fasteoi uhci_hcd:usb3 18: 61430 53863 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4 16: 103141 87919 IO-APIC-fasteoi uhci_hcd:usb3 18: 66678 58625 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4 ------- irq 16: nobody cared, Disabling IRQ #16. 16: 107840 92161 IO-APIC-fasteoi uhci_hcd:usb3 18: 71854 63458 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4 16: 107840 92161 IO-APIC-fasteoi uhci_hcd:usb3 18: 76498 68823 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb5, uhci_hcd:usb8 21: 1389 1539 IO-APIC-fasteoi uhci_hcd:usb4

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[alsa-devel] snd_pcm_hw_params_set_period_time_near
by Guilherme Raymo Longo 24 Jul '09

24 Jul '09

Mates, I can´t find the real meaning of this function. snd_pcm_hw_params_set_period_time_near At first I thought that it was used to set the time between 2 consecutive samples, but it is not. Does anyone could gimme some help with that? Tks

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 24 Jul '09

24 Jul '09

The following changes since commit 416c8fe3cdcd2d46a6ca12d8ec6143c62d8569d9: Takashi Iwai (1): ASoC: Kill direct accesses to driver_data are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Candelaria Villareal, Jorge (1): ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6 Chaithrika U S (1): ASoC: tlv320aic3x: Enable PLL when not bypassed Janusz Krzysztofik (2): ASoC: Add support for Conexant CX20442-11 voice modem codec ASoC: Jack handling enhancements as suggested by subsystem maintainer John Bonesio (1): ASoC: MPC5200: Increase the delay time between resets Joonyoung Shim (2): ASoC: MAX9877: separate callback functions ASoC: MAX9877: fix write operation for register Lopez Cruz, Misael (1): ASoC: OMAP: Staticise pcm creation function of omap-pcm Marek Vasut (2): ALSA: Allow passing platform_data to devices attached to AC97 bus ASoC: Switch palm27x-asoc to jack detection api Mark Brown (4): ASoC: Bodge around GCC 4.4.0 flow analysis bug in GCC 4.4.0 ASoC: Fix FLL reference clock division setup in WM8993 ASoC: Fix checkpatch issues in AD1938 Merge branch 'for-2.6.31' into for-2.6.32 include/sound/ac97_codec.h | 6 + include/sound/soc-dai.h | 1 + sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1938.c | 156 ++++++++-------- sound/soc/codecs/cx20442.c | 395 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cx20442.h | 19 ++ sound/soc/codecs/max9877.c | 110 +++++++---- sound/soc/codecs/tlv320aic3x.c | 11 +- sound/soc/codecs/wm8400.c | 9 +- sound/soc/codecs/wm8993.c | 2 + sound/soc/fsl/mpc5200_psc_ac97.c | 3 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/omap/sdp3430.c | 9 + sound/soc/pxa/palm27x.c | 204 +++++++------------- sound/soc/soc-core.c | 6 +- sound/soc/soc-jack.c | 20 ++- 17 files changed, 703 insertions(+), 255 deletions(-) create mode 100644 sound/soc/codecs/cx20442.c create mode 100644 sound/soc/codecs/cx20442.h

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