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

[alsa-devel] Report by PulseAudio: `alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail.`
by Paul Menzel 29 Sep '11

29 Sep '11

Dear ALSA developers, I got the following messages in `/var/log/syslog`. Aug 24 09:29:05 doe pulseaudio[2735]: core-util.c: Home directory /etc/timidity not ours. Aug 24 09:29:05 doe pulseaudio[2735]: lock-autospawn.c: Cannot access autospawn lock. Aug 24 09:29:05 doe pulseaudio[2735]: main.c: Failed to acquire autospawn lock Aug 24 13:00:33 doe pulseaudio[3118]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Aug 24 13:00:33 doe pulseaudio[3118]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 24 13:00:33 doe pulseaudio[3118]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Additionally I am seeing the a lot of the following messages in the kernel ring buffer (`dmesg`) [14293.422591] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422612] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422633] hda-intel: spurious response 0xe0160:0x0, last cmd=0x1f0004 [14293.422653] hda-intel: spurious response 0x1:0x0, last cmd=0x1f0004 [14293.422675] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422695] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422716] hda-intel: spurious response 0x20025:0x0, last cmd=0x1f0004 but I do not know if this is related. I am using the Debian Linux kernel 2.6.32 which includes all longterm patches up to 2.6.32.40. So only the ALSA drivers version 1.0.21 are used. Is there a commit that addresses the above behavior and which could be backported? Please find the output of `alsa-info.sh` [1] at the end. Thanks, Paul [1] http://alsa-project.org/main/index.php/Help_To_Debug !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Wed Aug 24 11:24:45 UTC 2011 !!Linux Distribution !!------------------ Debian GNU/Linux wheezy/sid \n \l !!DMI Information !!--------------- Manufacturer: System manufacturer Product Name: System Product Name Product Version: System Version !!Kernel Information !!------------------ Kernel release: 2.6.32-5-amd64 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.21 Library version: 1.0.24.1 Utilities version: 1.0.24.2 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [VT82xx ]: HDA-Intel - HDA VIA VT82xx HDA VIA VT82xx at 0xfbffc000 irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 20:01.0 Audio device: VIA Technologies, Inc. VT1708/A [Azalia HDAC] (VIA High Definition Audio Controller) (rev 10) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- 20:01.0 0403: 1106:3288 (rev 10) Subsystem: 1043:8290 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> patch : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC662 rev1 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0662 Subsystem Id: 0x10438290 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A Invalid AFG subtree --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 9 Aug 24 09:27 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 8 Aug 24 09:27 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 7 Aug 24 09:27 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 6 Aug 24 13:07 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 5 Aug 24 09:27 /dev/snd/pcmC0D1c crw-rw----+ 1 root audio 116, 4 Aug 24 09:27 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 3 Aug 24 09:27 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 Aug 24 09:27 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Aug 24 09:27 . drwxr-xr-x 3 root root 220 Aug 24 09:27 .. lrwxrwxrwx 1 root root 12 Aug 24 09:27 pci-0000:20:01.0 -> ../controlC0 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) pcm.pulse { type pulse } ctl.pulse { type pulse } pcm.!default { type pulse } ctl.!default { type pulse } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: VT82xx [HDA VIA VT82xx], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: VT82xx [HDA VIA VT82xx], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: VT82xx [HDA VIA VT82xx], device 0: ALC662 rev1 Analog [ALC662 rev1 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: VT82xx [HDA VIA VT82xx], device 1: ALC662 rev1 Digital [ALC662 rev1 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [VT82xx] Card hw:0 'VT82xx'/'HDA VIA VT82xx at 0xfbffc000 irq 17' Mixer name : 'Realtek ALC662 rev1' Components : 'HDA:10ec0662,10438290,00100101' Controls : 37 Simple ctrls : 20 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 63 [98%] [-1.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pswitch penum Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 254 [100%] [0.20dB] Front Right: Playback 254 [100%] [0.20dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 20 [65%] [-4.50dB] [on] Front Right: Playback 20 [65%] [-4.50dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 16 [52%] [-10.50dB] [on] Front Right: Playback 16 [52%] [-10.50dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 17 [55%] [-9.00dB] [on] Front Right: Playback 17 [55%] [-9.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 17 [55%] [-9.00dB] [on] Front Right: Playback 17 [55%] [-9.00dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined cswitch cswitch-joined penum Playback channels: Mono Capture channels: Mono Mono: Playback [on] Capture [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 24 [77%] [22.50dB] [on] Front Right: Capture 24 [77%] [22.50dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 9 [29%] [0.00dB] [off] Front Right: Capture 9 [29%] [0.00dB] [off] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '2ch' Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 66 [55%] [3.00dB] Front Right: Capture 66 [55%] [3.00dB] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' !!Alsactl output !!------------- --startcollapse-- state.VT82xx { control.1 { iface MIXER name 'Front Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.2 { iface MIXER name 'Front Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.3 { iface MIXER name 'Surround Playback Volume' value.0 64 value.1 64 comment { access 'read write' type INTEGER count 2 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.4 { iface MIXER name 'Surround Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.5 { iface MIXER name 'Center Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.6 { iface MIXER name 'LFE Playback Volume' value 64 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 0 } } control.7 { iface MIXER name 'Center Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.8 { iface MIXER name 'LFE Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.9 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.10 { iface MIXER name 'CD Playback Volume' value.0 17 value.1 17 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -900 dbvalue.1 -900 } } control.11 { iface MIXER name 'CD Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.12 { iface MIXER name 'Line Playback Volume' value.0 16 value.1 16 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -1050 dbvalue.1 -1050 } } control.13 { iface MIXER name 'Line Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.14 { iface MIXER name 'Mic Playback Volume' value.0 17 value.1 17 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -900 dbvalue.1 -900 } } control.15 { iface MIXER name 'Mic Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.16 { iface MIXER name 'Front Mic Playback Volume' value.0 20 value.1 20 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -450 dbvalue.1 -450 } } control.17 { iface MIXER name 'Front Mic Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.18 { iface MIXER name 'Channel Mode' value '2ch' comment { access 'read write' type ENUMERATED count 1 item.0 '2ch' item.1 '6ch' } } control.19 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.20 { iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.21 { iface MIXER name 'Capture Volume' value.0 24 value.1 24 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 2250 dbvalue.1 2250 } } control.22 { iface MIXER name 'Capture Volume' index 1 value.0 9 value.1 9 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -1350 dbmax 3300 dbvalue.0 0 dbvalue.1 0 } } control.23 { iface MIXER name 'Input Source' value Mic comment { access 'read write' type ENUMERATED count 1 item.0 Mic item.1 'Front Mic' item.2 Line item.3 CD } } control.24 { iface MIXER name 'Input Source' index 1 value Mic comment { access 'read write' type ENUMERATED count 1 item.0 Mic item.1 'Front Mic' item.2 Line item.3 CD } } control.25 { iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.26 { iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.27 { iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access 'read write' type IEC958 count 1 } } control.28 { iface MIXER name 'IEC958 Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.29 { iface MIXER name 'IEC958 Default PCM Playback Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.30 { iface MIXER name 'IEC958 Capture Switch' value false comment { access 'read write' type BOOLEAN count 1 } } control.31 { iface MIXER name 'IEC958 Capture Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' comment { access read type IEC958 count 1 } } control.32 { iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -3450 dbvalue.1 -3450 } } control.33 { iface MIXER name 'Beep Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.34 { iface MIXER name 'Master Playback Volume' value 63 comment { access 'read write' type INTEGER count 1 range '0 - 64' dbmin -6400 dbmax 0 dbvalue.0 -100 } } control.35 { iface MIXER name 'Master Playback Switch' value true comment { access 'read write' type BOOLEAN count 1 } } control.36 { iface MIXER name 'PCM Playback Volume' value.0 254 value.1 254 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 -20 dbvalue.1 -20 } } control.37 { iface MIXER name 'Digital Capture Volume' value.0 66 value.1 66 comment { access 'read write user' type INTEGER count 2 range '0 - 120' tlv '0000000100000008fffff44800000032' dbmin -3000 dbmax 3000 dbvalue.0 300 dbvalue.1 300 } } } --endcollapse-- !!All Loaded Modules !!------------------ Module binfmt_misc via drm ip6table_filter ip6_tables iptable_filter ip_tables x_tables powernow_k8 cpufreq_conservative cpufreq_userspace cpufreq_stats cpufreq_powersave reiserfs xfs exportfs fuse kvm_amd kvm loop firewire_sbp2 firewire_core crc_itu_t snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_hwdep snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_midi snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device amd64_edac_mod snd psmouse evdev edac_core parport_pc soundcore i2c_viapro parport snd_page_alloc k8temp asus_atk0110 shpchp edac_mce_amd serio_raw i2c_core pci_hotplug processor button ext3 jbd mbcache sha256_generic aes_x86_64 aes_generic cbc dm_crypt dm_mod raid1 md_mod sg usbhid hid sr_mod cdrom sd_mod crc_t10dif ata_generic uhci_hcd pata_via ehci_hcd sata_via libata thermal thermal_sys via_rhine mii usbcore nls_base scsi_mod !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x01014010 0x15 0x411111f0 0x16 0x411111f0 0x18 0x01a19840 0x19 0x02a19950 0x1a 0x0181304f 0x1b 0x02214120 0x1c 0x593301f0 0x1d 0x4005c603 0x1e 0x99430130 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: !!ALSA/HDA dmesg !!------------------ [ 38.078771] alloc kstat_irqs on node 0 [ 38.078780] HDA Intel 0000:20:01.0: PCI INT A -> GSI 17 (level, low) -> IRQ 17 [ 38.078848] HDA Intel 0000:20:01.0: setting latency timer to 64 [ 38.078852] HDA Intel 0000:20:01.0: PCI: Disallowing DAC for device [ 38.165465] input: HDA Digital PCBeep as /devices/pci0000:20/0000:20:01.0/input/input5 [ 38.229088] input: ImPS/2 Logitech Wheel Mouse as /devices/platform/i8042/serio1/input/input6 -- [ 146.194407] ioctl32(Xorg:2515): Unknown cmd fd(11) cmd(c0246400){t:'d';sz:36} arg(08665f10) on /dev/dri/card0 [ 185.172076] hda-intel: IRQ timing workaround is activated for card #0. Suggest a bigger bdl_pos_adj. [ 327.111398] eth0: link up, 100Mbps, full-duplex, lpa 0x45E1 [ 337.652026] eth0: no IPv6 routers present [14293.421429] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421447] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421467] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421488] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421509] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421530] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421550] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421571] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421592] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421613] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421634] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421655] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421677] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421695] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421716] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421737] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421758] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421778] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421799] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421820] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421841] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421861] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421882] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421903] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421924] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421945] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421965] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.421987] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422007] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422028] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422049] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422070] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422090] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422112] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422132] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422153] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422174] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422195] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422216] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422236] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422257] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422280] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422300] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422321] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422342] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422363] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422384] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422405] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422425] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422446] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422467] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422488] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422507] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422528] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422549] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422570] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422591] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422612] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422633] hda-intel: spurious response 0xe0160:0x0, last cmd=0x1f0004 [14293.422653] hda-intel: spurious response 0x1:0x0, last cmd=0x1f0004 [14293.422675] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422695] hda-intel: spurious response 0x0:0x0, last cmd=0x1f0004 [14293.422716] hda-intel: spurious response 0x20025:0x0, last cmd=0x1f0004

3 16

[alsa-devel] dmix problems with analog out (HDMI out works fine with dmix)
by Robert Krakora 29 Sep '11

29 Sep '11

Hi All, I am using ALSA 1.0.24 with the attached asound.conf. I am attempting to share (mix) the analog output between Asterisk and aplay. However, when Asterisk is running, my application throws an error when attempting to play a wave file via aplay. If I stop Asterisk, aplay plays the wave file just fine. If I remove the asound.conf file, Asterisk and aplay are both happy and share the analog output just find with the default ALSA dmix implementation. However, we need this asound.conf to work for us for other reasons. What is wrong with it. I have even tried replacing dmix with dshare and get the same bad behavior. We are running CentOS 6.0. As a caveat, I am attempting to use dmix with analog audio. If I use it with HDMI audio, it works flawlessly (same asound.conf with tweek to specify HDMI). Thanks in advance for your help. [root@devkrakorar snd]# su silentm -c "aplay -Dplug:dmixer /usr/share/sounds/alsa/Front_ Center.wav" ALSA lib pcm_hw.c:1293:(snd_pcm_hw_open) open '/dev/snd/pcmC0D0p' failed (-22): Invalid argument ALSA lib pcm_dmix.c:1068:(snd_pcm_dmix_open) unable to open slave aplay: main:660: audio open error: Invalid argument [root@devkrakorar snd]# Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext 212 (317)663-0808 Fax

2 7

[alsa-devel] [PATCH v2 0/4] ASoC: twl6040: Gain ramp code cleanups
by Peter Ujfalusi 29 Sep '11

29 Sep '11

Hello, Changes since v1: - Loop counters cleaned, commented in patch 2 - Use cancel_delayed_work_sync to make sure that we do not have pending work, or executing work when modifying the variables used by the ramp code. - patches already taken left out. Intro from v1: the following series cleans up the gain ramp code found in the twl6040 codec driver. Main changes: - use one workqueue for the twl6040 codec driver (instead of the original 3) - Delays between the steps does not need to be different among the range. I assume, that the original code contained copy-paste snippets from wm8350 for this part - Cleanups for the DAPM_OUT_DRV_E event handler code. - delayed_works moved within their corresponding struct. The series has been generated on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch + ASoC: omap-mcpdm/twl6040: Offset cancellation series. Side note: I have done this to better understand (while cleaning up the twl6040 driver) the requirements for the ramp code, and to study the possibility of adding support in the core for this (to handle the wm8350, and twl6040 in a generic way later). I'm still looking at the optimal implementation without ending up with too complicated code/structures... Regards, Peter --- Peter Ujfalusi (4): ASoC: twl6040: One workqueue should be enough ASoC: twl6040: correct loop counters for HS/HF ramp code ASoC: twl6040: Shift 2 identifies the HS output in out_drv_event ASoC: twl6040: Simplify code in out_drv_event for pending work check sound/soc/codecs/twl6040.c | 68 +++++++++++++------------------------------ 1 files changed, 21 insertions(+), 47 deletions(-) -- 1.7.6.1

1 5

[alsa-devel] alsa_aplay doesn't work on android on hdk7108
by Hou-xiang ZHU 29 Sep '11

29 Sep '11

Hello, I am trying to add alsa feature to Android on hdk7108 (which is sh4 architecture, not arm). Basically on non android linux platform ,i.e STLinux-2.4, the driver and alsa-libs and alsa-utils works with the following test command: --------------------------- #aplay -v -t raw -D hw:1,0 -c 2 -r 32000 -f S16_BE /32K_16Bit_Stereo.pcm" --------------------------- But when working on Android platform, because the alsa-lib/alsa-utils are different from STLinux-2.4, so we ask someone the two packages from a arm platform. and compilation ok. but during run alsa_aplay, the following error occurred: ----------------------------------- #/system/bin/alsa_aplay -v -t raw -D hw:1,0 -c 2 -r 32000 -f S16_BE /32K_16Bit_Stereo.pcm ALSA lib vendor/STMicroelectronics/hdk7108/alsa/alsa-lib/src/confmisc.c:137:(snd_config_get_bool) confmisc.c in func snd_config_get_bool line 137 Invalid type for nonblock ALSA lib vendor/STMicroelectronics/hdk7108/alsa/alsa-lib/src/pcm/pcm_hw.c:1417:(_snd_pcm_hw_open) Invalid type for subdevice aplay: main:656: in aplay.c func main line 656 audio open error: Invalid argument aplay/main.c in func main line 452 alsa-lib/src/pcm/pcm.c func snd_pcm_open line 2242 after call snd_config_update(), err=1 ---------------------------------- looks like alsa configuration issue. I noticed that on STLinux-2.4 platform, the alsa-libs is 1.0.21a and on android platform, it is 1.0.16, can this be a issue? or configuration file wrong or else? how to debug this? Thanks you very much! Houxiang

1 0

[alsa-devel] [PATCH] fix the bug for WM8962 DSP_A and DSP_B settings
by Susan Gao 29 Sep '11

29 Sep '11

Signed-off-by: Susan Gao <sgao(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 117349b..49ea487 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3030,8 +3030,7 @@ static int wm8962_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - aif0 |= WM8962_LRCLK_INV; - case SND_SOC_DAIFMT_DSP_B: aif0 |= 3; + case SND_SOC_DAIFMT_DSP_B: + aif0 |= 3 | WM8962_LRCLK_INV; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: tlv320dac33: Add guarding parentheses to macros
by Axel Lin 29 Sep '11

29 Sep '11

Put parentheses around macro argument uses. This avoids pitfalls for the programmer, where the argument expansion does not give the expected result, for example: SAMPLES_TO_US(substream->runtime->rate, dac33->uthr - DAC33_MODE7_MARGIN + 1); Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/tlv320dac33.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 243d177..389b1c2 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -55,13 +55,13 @@ #define BURST_BASEFREQ_HZ 49152000 #define SAMPLES_TO_US(rate, samples) \ - (1000000000 / ((rate * 1000) / samples)) + (1000000000 / (((rate) * 1000) / (samples))) #define US_TO_SAMPLES(rate, us) \ - (rate / (1000000 / (us < 1000000 ? us : 1000000))) + ((rate) / (1000000 / ((us) < 1000000 ? (us) : 1000000))) #define UTHR_FROM_PERIOD_SIZE(samples, playrate, burstrate) \ - ((samples * 5000) / ((burstrate * 5000) / (burstrate - playrate))) + (((samples)*5000) / (((burstrate)*5000) / ((burstrate) - (playrate)))) static void dac33_calculate_times(struct snd_pcm_substream *substream); static int dac33_prepare_chip(struct snd_pcm_substream *substream); -- 1.7.4.1

3 2

[alsa-devel] [PATCH 1/6] ASoC: Remove unused "control_data" field of struct ak4671_priv
by Axel Lin 29 Sep '11

29 Sep '11

The control_data field is used to initialize the codec's control_data field, but since this is also done by the snd-soc-cache core, the redundant assignment can be removed and the field can be dropped. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/ak4671.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 88b29f8..2ecf128 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -26,7 +26,6 @@ /* codec private data */ struct ak4671_priv { enum snd_soc_control_type control_type; - void *control_data; }; /* ak4671 register cache & default register settings */ @@ -675,7 +674,6 @@ static int __devinit ak4671_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, ak4671); - ak4671->control_data = client; ak4671->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, -- 1.7.4.1

3 8

[alsa-devel] [patch] sound: oss: use strlcpy() in sound_timer_init()
by Dan Carpenter 29 Sep '11

29 Sep '11

sound_timer.info.name is a 32 character buffer. This function only has one caller (in sound/oss/ad1848.c) and it passes as 128 character buffer as "name". I don't know if this is a problem in real life, and I doubt we're going to add more OSS drivers so it's unlikely to become an issue. But we may as well take care of it. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index 48cda6c..8021c85 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -320,7 +320,7 @@ void sound_timer_init(struct sound_lowlev_timer *t, char *name) n = sound_alloc_timerdev(); if (n == -1) n = 0; /* Overwrite the system timer */ - strcpy(sound_timer.info.name, name); + strlcpy(sound_timer.info.name, name, sizeof(sound_timer.info.name)); sound_timer_devs[n] = &sound_timer; } EXPORT_SYMBOL(sound_timer_init);

2 1

Re: [alsa-devel] snd_pcm_pause kills plaback handle on OMAP4
by john＠johnandjuliet.com 28 Sep '11

28 Sep '11

I added the patch and no go... I added some printks and it is not even being called. I am now suspecting the mcpdm side because when I am doing a pause/resume on the capture handle that the McBSP does become active and is spitting out white-noise-like samples but no audio. So that may mean the mcpdm or the 6040 is the problem??? JBP > -------- Original Message -------- > Subject: Re: RE: [alsa-devel] snd_pcm_pause kills plaback handle on > OMAP4 > From: Péter Ujfalusi <peter.ujfalusi(a)ti.com> > Date: Tue, September 27, 2011 2:37 am > To: john(a)johnandjuliet.com > Cc: alsa-devel(a)alsa-project.org > > > Hi, > > Can you try this patch, if it helps: > http://mailman.alsa-project.org/pipermail/alsa-devel/2011- > September/044227.html > > -- > Péter > > On Friday 16 September 2011 06:31:09 john(a)johnandjuliet.com wrote: > > Thanks for the response. > > > > Yea I guess my question is a bit thin and the setup is a bit more > > complex so I should list the particulars: > > > > * This is on an OMAP4 BLAZE ES2.2 or 2.3 processor > > * Running Android Linux 2.6.35 from the TI release version G.1 > > * I am actually configuring ALSA to utilize the MODEM audio path > > (which is hw:0,5 that routes McBSP2 audio to/from the 6040's speaker and > > mic.) > > > > > > * The McBSP is configured for slave mode network SSI with the audio flow > > on slots 0 and 1 > > (the default was I2S) > > * McBSP is configured for 8K sps two channel 32-bit samples (though it > > could be 16 bit samples too) > > > > I guess the way I wrote the first email that I made it sound like it is > > the McBSP that is not working > > however it could be that while I am sending in audio perhaps the McBSP2 > > is working (not sure where I could look at data comming in down in the > > kernel <help plz> but if the McBSP2 is working then it could be that the > > data is not getting to the 6040?! > > > > > On Wednesday 14 September 2011 11:13:22 john(a)johnandjuliet.com wrote: > > > > I'm trying to control the McBSP2 that is used for the modem in an > > > > android environment (BLAZE OMAP4) > > > > > > What kernel are you using? > > > > 2.6.35 released by TI for Android Gingerbread. TI tags this as their G1 > > release (they are up to G3 now) > > > > > > What I am trying to do is control the audio paths for capture and > > > > playback seperatly like in a SIMPLEX (half duplex) instead of full > > > > DUPLEX mode audio .. > > > > > > So either playback, or capture, but never both at the same time? > > > > Yes. The device cannot do full duplex. The ALSA config for hw:0,5 is > > to route audio from McBSP2 to/from the 6040 (ABE) > > > > The stock Android alsa lib package opens up handles for both capture and > > playback and then turns on audio hardware both MICs and Speakers. > > > > > > I also want to stop the audio without total > > > > shutdown of the audio system. > > > > > > > > From what I can see that would be to call snd_pcm_pause(handle,bool) > > > > > > > > But here is what happens. With the handle forthe plaback path > > > > everything works OK... But if I call that then the capture path > > > > stops > > > > working... even if I try to turn on (unpause) both paths. > > > > > > So you have both playback and capture opened, if you pause the playback, > > > and release it, then the capture path stops working? > > > Or you mean that doing the pause/release with playback, then closing it, > > > after that the capture is not working? > > > > If they are both flowing and I pause capture then playback stops and > > though I can un-pause capture playback never comes back. > > > > I'll have to double check but I think if I pause playback and then > > un-pause it... that too breaks and playback never resumes. > > > > > > This implies some sort of bug in the omap soc driver? > > > > > > Everything is possible. I have not heard such about such a problem > > > myself. If this issue exist on OMAP4, it should be present on OMAP3 as > > > well... > > OK. perhaps though as I just noted I left out that this could actually > > be a problem on the speaker flow out and not the McBSP2 flow in.. What > > do you think? The OMAP4 audio system is much different than OMAP3 on > > the speaker/mic side so I guess it could be a problem there? > > > > > Could you be more specific on the scenario? > > > What codec you have attached to McBSP2 port? > > > Are you using ABE, or are you using the McBSP2 port directly? > > > > I have my own simplex modem connected and it is generating 8ksps audio > > on slot 0 (first 32 bits) and the McBSP is configured that way. > > > > I am using the ABE to turn on the speaker hardware and the 6040 side of > > things. > > > > Mostly I am using the TI ALSA layer they provide for Android ALSA audio > > control and simply placed into it at a key point a snd_pcm_pause() for > > when I want to turn the capture off or on and another place for the > > playback to be off or on. > > > > Typically capture and playback are not both on at the same time.. You > > are either speaking or you are listening. This is like a CB radio or a > > Walkie Talkie type of situation. To save power and also reduce static > > the speaker is only turned on when the modem on McBSP2 indicates that it > > wants to play audio and that is when I would do a > > snd_pcm_pause(playbackhandle,0); when the audio from the modem stops and > > I get the signal from the modem then I call > > snd_pcm_pause(playbackhandle,1); > > > > When I want to talk then I do the snd_pcm_pause(capture_handle,0); and > > when I stop talking then I do the pause on the capture_handle. > > > > I will have to check but I think if I pause EITHER one then playback > > stops working. Playback does work if all I do is an open and not mess > > with it and just do the ALSA set controls to turn on the speaker. > > > > > CC-ing Jarkko to catch his attention... > > > > > > -- > > > Péter > > > > JBP

1 0

[alsa-devel] [PATCH] ASoC: Instantiate DAPM widgets before we do the DAI link init
by Mark Brown 28 Sep '11

28 Sep '11

The DAI init function may want to do something that needs the widgets to be instantiated. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a58c1fc..1ed8093 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1059,6 +1059,9 @@ static int soc_post_component_init(struct snd_soc_card *card, temp = codec->name_prefix; codec->name_prefix = NULL; + /* Make sure all DAPM widgets are instantiated */ + snd_soc_dapm_new_widgets(&codec->dapm); + /* do machine specific initialization */ if (!dailess && dai_link->init) ret = dai_link->init(rtd); @@ -1070,9 +1073,6 @@ static int soc_post_component_init(struct snd_soc_card *card, } codec->name_prefix = temp; - /* Make sure all DAPM widgets are instantiated */ - snd_soc_dapm_new_widgets(&codec->dapm); - /* register the rtd device */ rtd->codec = codec; rtd->dev.parent = card->dev; -- 1.7.6.3

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