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[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

1 0

[alsa-devel] [PATCH] ASoC: Also count neighbour checks for supplies
by Mark Brown 28 Sep '11

28 Sep '11

Missed when the stat was originally added. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5434767..12bd01a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -828,6 +828,8 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { + DAPM_UPDATE_STAT(w, neighbour_checks); + if (path->weak) continue; -- 1.7.6.3

1 0

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

28 Sep '11

OK... I added that patch. And I added some printks inside it to see if it executes that code and it never does. Oh well. 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] Signaling user-space through kcontrol
by Patrick Lai 28 Sep '11

28 Sep '11

Hi Mark, The codec I am working on, beside headset detection, also have other type of events to be propagated to user space. Per our discussion at IPC, there is way to signal user-space through kcontrol. Is it through calling snd_ctl_notify()? What is the ALSA library API to intercept the event? Is there an sample user-space code which gets kcontrol event? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

5 9

Re: [alsa-devel] dmix
by Robert Krakora 28 Sep '11

28 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 ~]# aplay /usr/share/sounds/alsa/Front_Center.wav pcm_name = default ALSA lib pcm_dmix.c:1068:(snd_pcm_dmix_open) unable to open slave aplay: main:662: audio open error: Invalid argument Best Regards, -- Rob Krakora MessageNet Systems 101 East Carmel Dr. Suite 105 Carmel, IN 46032 (317)566-1677 Ext 212 (317)663-0808 Fax

1 0

[alsa-devel] [PATCH v3] ALSA: hdmi: expose ELD control
by Pierre-Louis Bossart 28 Sep '11

28 Sep '11

Applications may want to read ELD information to understand what codecs are supported on the HDMI receiver and handle the a-v delay for better lip-sync. ELD information is exposed in a device-specific IFACE_PCM kcontrol. Tested both with amixer and PulseAudio. ELD control size is set to zero in case of errors or wrong configurations. No notifications are implemented for now, it is expected that jack detection is used to reconfigure the audio outputs. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 13 ++++-- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_hdmi.c | 88 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 98 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index c34f730..f1c621d 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -318,6 +318,11 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, int size; unsigned char *buf; + /* + * ELD size is initialized to zero in caller function. If no errors and + * ELD is valid, actual eld_size is assigned in hdmi_update_eld() + */ + if (!eld->eld_valid) return -ENOENT; @@ -327,14 +332,13 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); size = 128; } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE) { snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); return -ERANGE; } - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; + /* set ELD buffer */ + buf = eld->eld_buffer; for (i = 0; i < size; i++) { unsigned int val = hdmi_get_eld_data(codec, nid, i); @@ -356,7 +360,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, ret = hdmi_update_eld(eld, buf, size); error: - kfree(buf); return ret; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index aaefa7c..04d730f 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -621,6 +621,7 @@ struct cea_sad { }; #define ELD_FIXED_BYTES 20 +#define ELD_MAX_SIZE 256 #define ELD_MAX_MNL 16 #define ELD_MAX_SAD 16 @@ -645,6 +646,7 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3f1f6ac..177b4af 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -324,6 +324,86 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) return -EINVAL; } +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + + spec = codec->spec; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + + pin_idx = kcontrol->private_value; + if (snd_BUG_ON(pin_idx < 0)) + return -EINVAL; + + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + uinfo->count = eld->eld_size; + + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + int size; + + spec = codec->spec; + pin_idx = kcontrol->private_value; + + if (snd_BUG_ON(pin_idx < 0)) + return -EINVAL; + + memset(ucontrol->value.bytes.data, 0, ELD_MAX_SIZE); + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + size = eld->eld_size; + + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, size); + + return 0; +} + +static struct snd_kcontrol_new eld_bytes_ctl = { + .access = SNDRV_CTL_ELEM_ACCESS_READ | SNDRV_CTL_ELEM_ACCESS_VOLATILE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, +}; + +static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, + int device) +{ + struct snd_kcontrol *kctl; + struct hdmi_spec *spec = codec->spec; + int err; + + if (pin_idx < 0) + return -EINVAL; + + kctl = snd_ctl_new1(&eld_bytes_ctl, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = pin_idx; + kctl->id.device = device; + + err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl); + if (err < 0) + return err; + + return 0; +} + #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) @@ -1193,6 +1273,14 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pin_idx); + + /* add control for ELD Bytes */ + err = hdmi_create_eld_ctl(codec, + pin_idx, + spec->pcm_rec[pin_idx].device); + + if (err < 0) + return err; } return 0; -- 1.7.6.2

2 1

[alsa-devel] "modprobe snd_hda_intel" crashes on my hardware
by Abdelghani Ouchabane 28 Sep '11

28 Sep '11

upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Mon Aug 22 22:33:45 UTC 2011 !!Linux Distribution !!------------------ Fedora release 15 (Lovelock) Fedora release 15 (Lovelock) Fedora release 15 (Lovelock) Fedora release 15 (Lovelock) !!DMI Information !!--------------- Manufacturer: Product Name: Product Version: !!Kernel Information !!------------------ Kernel release: 2.6.40.3-0.119.delos.i686 Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: No !!ALSA Version !!------------ Driver version: 1.0.24 Library version: Utilities version: 1.0.24.1 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- No sound servers found. !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfea38000 irq 10 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:27d8 (rev 02) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 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 beep_mode : 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 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 : -1 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 : 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 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: VIA VT1708 Address: 2 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x11061708 Subsystem Id: 0xffffffff Revision Id: 0x100700 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 GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x10 [Audio Output] wcaps 0x411: Stereo Device: name="VT1708 Analog", type="Audio", device=0 Converter: stream=0, channel=0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 Power: setting=D0, actual=D0 Node 0x13 [Audio Output] wcaps 0x411: Stereo Converter: stream=0, channel=0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 Power: setting=D0, actual=D0 Node 0x14 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xa]: 16 24 formats [0x5]: PCM AC3 Node 0x15 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="VT1708 Analog", type="Audio", device=0 Amp-In caps: ofs=0x00, nsteps=0x14, stepsize=0x06, mute=1 Amp-In vals: [0x0f 0x0f] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x440]: 48000 192000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 Power: setting=D0, actual=D0 Connection: 1 0x18 Node 0x16 [Audio Input] wcaps 0x100311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Validity Digital category: 0x0 PCM: rates [0x1f0]: 32000 44100 48000 88200 96000 bits [0xa]: 16 24 formats [0x5]: PCM AC3 Connection: 1 0x26 Node 0x17 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Master Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Master Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x06, mute=1 Amp-In vals: [0x18 0x18] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 6 0x10 0x24 0x1d 0x1e 0x21 0x13 Node 0x18 [Audio Selector] wcaps 0x300101: Stereo Control: name="Input Source", index=0, device=0 Connection: 5 0x17* 0x24 0x1d 0x1e 0x21 Node 0x19 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1b, nsteps=0x1b, stepsize=0x06, mute=1 Amp-Out vals: [0x00 0x00] Connection: 1 0x11 Node 0x1a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1b, nsteps=0x1b, stepsize=0x06, mute=1 Amp-Out vals: [0x00 0x00] Connection: 1 0x12 Node 0x1b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1b, nsteps=0x1b, stepsize=0x06, mute=1 Amp-Out vals: [0x00 0x00] Connection: 1 0x13 Node 0x1c [Pin Complex] wcaps 0x40010d: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0000f000: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Other DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0x00: Connection: 1 0x19 Node 0x1d [Pin Complex] wcaps 0x400101: Stereo Control: name="Smart 5.1", index=0, device=0 Pincap 0x00000334: IN OUT Detect Vref caps: HIZ 50 Pin Default 0x00a0f040: [Jack] Mic at Ext N/A Conn = Unknown, Color = Other DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x41: OUT VREF_50 Connection: 1 0x1a Node 0x1e [Pin Complex] wcaps 0x400101: Stereo Pincap 0x00000334: IN OUT Detect Vref caps: HIZ 50 Pin Default 0xfffff7ff: [Both] Other at Oth UNKNOWN Conn = Other, Color = Other DefAssociation = 0xf, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Connection: 1 0x19 Node 0x1f [Pin Complex] wcaps 0x40010d: Stereo Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x1b, nsteps=0x1b, stepsize=0x06, mute=1 Amp-Out vals: [0x15 0x15] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0100f010: [Jack] Line Out at Ext Rear Conn = Unknown, Color = Other DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Connection: 1 0x17 Node 0x20 [Pin Complex] wcaps 0x40010d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x1b, nsteps=0x1b, stepsize=0x06, mute=1 Amp-Out vals: [0x1b 0x1b] Pincap 0x0000001c: OUT HP Detect Pin Default 0x0020f000: [Jack] HP Out at Ext N/A Conn = Unknown, Color = Other DefAssociation = 0x0, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Connection: 1 0x17 Node 0x21 [Pin Complex] wcaps 0x400101: Stereo Pincap 0x00000334: IN OUT Detect Vref caps: HIZ 50 Pin Default 0xffffffff: [Both] Other at Oth UNKNOWN Conn = Other, Color = Other DefAssociation = 0xf, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0x00: VREF_HIZ Connection: 1 0x1b Node 0x22 [Pin Complex] wcaps 0x400101: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x0000f002: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Other DefAssociation = 0x0, Sequence = 0x2 Pin-ctls: 0x00: Connection: 1 0x1a Node 0x23 [Pin Complex] wcaps 0x400101: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x0000f004: [Jack] Line Out at Ext N/A Conn = Unknown, Color = Other DefAssociation = 0x0, Sequence = 0x4 Pin-ctls: 0x00: Connection: 1 0x1b Node 0x24 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0xfffbffff: [Both] Other at Oth UNKNOWN Conn = Comb, Color = Other DefAssociation = 0xf, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0xffffffff: [Both] Other at Oth UNKNOWN Conn = Other, Color = Other DefAssociation = 0xf, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x14 Node 0x26 [Pin Complex] wcaps 0x400201: Stereo Digital Pincap 0x00010030: IN OUT EAPD EAPD 0x2: EAPD Pin Default 0xffffffff: [Both] Other at Oth UNKNOWN Conn = Other, Color = Other DefAssociation = 0xf, Sequence = 0xf Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Node 0x27 [Audio Input] wcaps 0x10051b: Stereo Amp-In Control: name="Capture Volume", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=1, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x14, stepsize=0x06, mute=1 Amp-In vals: [0x0f 0x0f] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x440]: 48000 192000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 Power: setting=D0, actual=D0 Connection: 1 0x21 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root cyclades 116, 5 Aug 23 00:33 /dev/snd/controlC0 crw-rw---- 1 root cyclades 116, 4 Aug 23 00:33 /dev/snd/hwC0D2 crw-rw---- 1 root cyclades 116, 3 Aug 23 00:33 /dev/snd/pcmC0D0c crw-rw---- 1 root cyclades 116, 2 Aug 23 00:33 /dev/snd/pcmC0D0p crw-rw---- 1 root cyclades 116, 1 Aug 23 00:33 /dev/snd/seq crw-rw---- 1 root cyclades 116, 33 Aug 23 00:33 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Aug 23 00:33 . drwxr-xr-x 3 root root 180 Aug 23 00:33 .. lrwxrwxrwx 1 root root 12 Aug 23 00:33 pci-0000:00:1b.0 -> ../controlC0 !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # # Place your global alsa-lib configuration here... # @hooks [ { func load files [ "/etc/alsa/pulse-default.conf" ] errors false } ] !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: VT1708 Analog [VT1708 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: VT1708 Analog [VT1708 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xfea38000 irq 10' Mixer name : 'VIA VT1708' Components : 'HDA:11061708,ffffffff,00100700' Controls : 15 Simple ctrls : 9 Simple mixer control 'Master Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 24 [77%] [1.75dB] [on] Front Right: Playback 24 [77%] [1.75dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 27 Mono: Front Left: Playback 27 [100%] [0.00dB] [on] Front Right: Playback 27 [100%] [0.00dB] [on] Simple mixer control 'Front',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 27 Mono: Front Left: Playback 21 [78%] [-10.50dB] [on] Front Right: Playback 21 [78%] [-10.50dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-40.25dB] [off] Front Right: Playback 0 [0%] [-40.25dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 20 Front Left: Capture 15 [75%] [26.25dB] [on] Front Right: Capture 15 [75%] [26.25dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 20 Front Left: Capture 15 [75%] [26.25dB] [on] Front Right: Capture 15 [75%] [26.25dB] [on] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Stereo Mixer' 'Mic' Item0: 'Stereo Mixer' Simple mixer control 'Jack Detect',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] Simple mixer control 'Smart 5.1',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [on] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { iface MIXER name 'Jack Detect' value true comment { access 'read write' type BOOLEAN count 1 } } control.2 { iface MIXER name 'Master Front Playback Volume' value.0 24 value.1 24 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -4025 dbmax 1400 dbvalue.0 175 dbvalue.1 175 } } control.3 { iface MIXER name 'Master Front Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.4 { iface MIXER name 'Front Playback Volume' value.0 21 value.1 21 comment { access 'read write' type INTEGER count 2 range '0 - 27' dbmin -4725 dbmax 0 dbvalue.0 -1050 dbvalue.1 -1050 } } control.5 { iface MIXER name 'Front Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.6 { iface MIXER name 'Headphone Playback Volume' value.0 27 value.1 27 comment { access 'read write' type INTEGER count 2 range '0 - 27' dbmin -4725 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } control.7 { iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.8 { iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -4025 dbmax 1400 dbvalue.0 -4025 dbvalue.1 -4025 } } control.9 { iface MIXER name 'Mic Playback Switch' value.0 false value.1 false comment { access 'read write' type BOOLEAN count 2 } } control.10 { iface MIXER name 'Smart 5.1' value true comment { access 'read write' type BOOLEAN count 1 } } control.11 { iface MIXER name 'Capture Volume' value.0 15 value.1 15 comment { access 'read write' type INTEGER count 2 range '0 - 20' dbmin 0 dbmax 3500 dbvalue.0 2625 dbvalue.1 2625 } } control.12 { iface MIXER name 'Capture Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.13 { iface MIXER name 'Capture Volume' index 1 value.0 15 value.1 15 comment { access 'read write' type INTEGER count 2 range '0 - 20' dbmin 0 dbmax 3500 dbvalue.0 2625 dbvalue.1 2625 } } control.14 { iface MIXER name 'Capture Switch' index 1 value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.15 { iface MIXER name 'Input Source' value 'Stereo Mixer' comment { access 'read write' type ENUMERATED count 1 item.0 'Stereo Mixer' item.1 Mic } } } --endcollapse-- !!All Loaded Modules !!------------------ Module snd_hda_codec_via snd_hda_intel snd_hda_codec snd_hwdep snd_seq snd_seq_device snd_pcm snd_timer snd soundcore snd_page_alloc ipv6 fakephp cgosdrv joydev mcs7830 usbnet mii usbtouchscreen iTCO_wdt iTCO_vendor_support i2c_i801 serio_raw pciehp xfs i915 drm_kms_helper drm i2c_algo_bit i2c_core video !!Sysfs Files !!----------- /sys/class/sound/hwC0D2/init_pin_configs: 0x1c 0x0000f000 0x1d 0x00a0f040 0x1e 0xfffff7ff 0x1f 0x0100f010 0x20 0x0020f000 0x21 0xffffffff 0x22 0x0000f002 0x23 0x0000f004 0x24 0xfffbffff 0x25 0xffffffff 0x26 0xffffffff /sys/class/sound/hwC0D2/driver_pin_configs: /sys/class/sound/hwC0D2/user_pin_configs: /sys/class/sound/hwC0D2/init_verbs: !!ALSA/HDA dmesg !!------------------ [ 43.978457] XFS (sda2): Ending clean mount [ 153.600185] HDA Intel 0000:00:1b.0: PCI INT A -> Link[LNKA] -> GSI 10 (level, low) -> IRQ 10 [ 153.600227] HDA Intel 0000:00:1b.0: setting latency timer to 64

5 25

[alsa-devel] [PATCH 1/2] ASoC: adau1701: Initialize codec->control_data before using it
by Axel Lin 28 Sep '11

28 Sep '11

Currently codec->control_data is not initialized before calling process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE). Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/adau1701.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 2758d5f..1354902 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -88,6 +88,7 @@ struct adau1701 { unsigned int dai_fmt; + void *control_data; }; static const struct snd_kcontrol_new adau1701_controls[] = { @@ -455,9 +456,11 @@ static struct snd_soc_dai_driver adau1701_dai = { static int adau1701_probe(struct snd_soc_codec *codec) { + struct adau1701 *adau1701 = snd_soc_codec_get_drvdata(codec); int ret; codec->dapm.idle_bias_off = 1; + codec->control_data = adau1701->control_data; ret = adau1701_load_firmware(codec); if (ret) @@ -500,6 +503,7 @@ static __devinit int adau1701_i2c_probe(struct i2c_client *client, return -ENOMEM; i2c_set_clientdata(client, adau1701); + adau1701->control_data = client; ret = snd_soc_register_codec(&client->dev, &adau1701_codec_drv, &adau1701_dai, 1); if (ret < 0) -- 1.7.4.1

3 3

[alsa-devel] [PATCH] ASoC: adau1701: Initialize codec->control_data before using it
by Axel Lin 28 Sep '11

28 Sep '11

Currently codec->control_data is not initialized before calling process_sigma_firmware(codec->control_data, ADAU1701_FIRMWARE). Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/adau1701.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index b7e4cc7..8b7e1c5 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -458,6 +458,7 @@ static int adau1701_probe(struct snd_soc_codec *codec) int ret; codec->dapm.idle_bias_off = 1; + codec->control_data = to_i2c_client(codec->dev); ret = adau1701_load_firmware(codec); if (ret) -- 1.7.4.1

2 2

[alsa-devel] [PATCH] ASoC: Fix setting adau1373_dai->master for SND_SOC_DAIFMT_CBS_CFS
by Axel Lin 28 Sep '11

28 Sep '11

In the case of SND_SOC_DAIFMT_CBS_CFS, adau1373_dai->master should be false. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/codecs/adau1373.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 2aa40c3..1ccf8dd 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -974,7 +974,7 @@ static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_CBS_CFS: ctrl = 0; - adau1373_dai->master = true; + adau1373_dai->master = false; break; default: return -EINVAL; -- 1.7.4.1

2 1

[alsa-devel] UCM and modem configurations
by Pierre-Louis Bossart 28 Sep '11

28 Sep '11

I've had a couple of questions internally on UCM and can't figure out how things would work with a cellular modem configured through AT commands (eg Intel/IFX XGOLD). For example UCM could be used to set the volumes and routing for a voice call, but to modify the acoustics processing on the modem one would still need to send a bunch of AT commands . So what's the best way to handle this case: - Is UCM only supposed to be used for non-cellular cases? - Do we need to wrap all the AT commands as ALSA controls? - Is there a way to abstract such commands with some UCM syntax? - Or do we need to implement another layer to abstract ALSA and AT commands (which essentially means using UCM concepts but not the UCM tools and configuration files)? Thanks for your feedback -Pierre

1 0

Re: [alsa-devel] [PATCH v2] ALSA: hdmi: expose ELD control
by Stephen Warren 27 Sep '11

27 Sep '11

Pierre-Louis Bossart wrote at Tuesday, September 27, 2011 9:49 AM: ... > > Set the access field to SNDRV_CTL_ELEM_ACCESS_READ. > > Also, unless we implement a notification, better to set > > SNDRV_CTL_ELEM_ACCESS_VOLATILE flag, too. > > The intent was to read the ELD after the device is opened, I don't think the > ELD can change during playback, the updates can only happen on a pin-sensing > event. I can't think of a good reason why a notification would be needed? If the user unplugs an HDMI monitor and plugs a different one in, and hence the X server turns off the output and re-enables it to cater for any required new display mode, would that force the audio client to close/re-open the audio device? I don't /think/ so. As such, I guess the ELD can change dynamically, so notification would be a good idea. -- nvpublic

2 2

[alsa-devel] [PATCH 0/6] ASoC: Remove unused control_data field from private struct
by Axel Lin 27 Sep '11

27 Sep '11

This serial removes unused control_data field from private struct. 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. Note: I don't have these hardware, only compile test on these patches. Axel Lin (6): ASoC: Remove unused "control_data" field of struct wm8904_priv ASoC: Remove unused "control_data" field of struct wm8940_priv ASoC: Remove unused "control_data" field of struct wm8960_priv ASoC: Remove unused "control_data" field of struct wm8978_priv ASoC: Remove unused "control_data" field of struct wm9081_priv ASoC: Remove unused "control_data" field of struct wm9090_priv sound/soc/codecs/wm8904.c | 2 -- sound/soc/codecs/wm8940.c | 3 --- sound/soc/codecs/wm8960.c | 3 --- sound/soc/codecs/wm8978.c | 3 --- sound/soc/codecs/wm9081.c | 3 --- sound/soc/codecs/wm9090.c | 3 --- 6 files changed, 0 insertions(+), 17 deletions(-) -- 1.7.4.1

2 7

[alsa-devel] [PATCH 1/3] mfd: Add WM1811 support
by Mark Brown 27 Sep '11

27 Sep '11

The WM1811 is mostly register compatible with the WM8994 and WM8958, providing a high performance audio hub CODEC in a small form factor suitable for ultra compact system designs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Due to the ASoC cross dependencies it seems like it'll be easier to merge this via ASoC - Samuel, does this make sense to you? drivers/mfd/wm8994-core.c | 28 ++++++++++++++++++++++++++++ include/linux/mfd/wm8994/core.h | 1 + 2 files changed, 29 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 719fee8..61894fc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -167,6 +167,18 @@ static struct mfd_cell wm8994_devs[] = { * and should be handled via the standard regulator API supply * management. */ +static const char *wm1811_main_supplies[] = { + "DBVDD1", + "DBVDD2", + "DBVDD3", + "DCVDD", + "AVDD1", + "AVDD2", + "CPVDD", + "SPKVDD1", + "SPKVDD2", +}; + static const char *wm8994_main_supplies[] = { "DBVDD", "DCVDD", @@ -227,6 +239,7 @@ static int wm8994_suspend(struct device *dev) switch (wm8994->type) { case WM8958: + case WM1811: ret = wm8994_reg_read(wm8994, WM8958_MIC_DETECT_1); if (ret < 0) { dev_err(dev, "Failed to read power status: %d\n", ret); @@ -375,6 +388,9 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) } switch (wm8994->type) { + case WM1811: + wm8994->num_supplies = ARRAY_SIZE(wm1811_main_supplies); + break; case WM8994: wm8994->num_supplies = ARRAY_SIZE(wm8994_main_supplies); break; @@ -395,6 +411,10 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) } switch (wm8994->type) { + case WM1811: + for (i = 0; i < ARRAY_SIZE(wm1811_main_supplies); i++) + wm8994->supplies[i].supply = wm1811_main_supplies[i]; + break; case WM8994: for (i = 0; i < ARRAY_SIZE(wm8994_main_supplies); i++) wm8994->supplies[i].supply = wm8994_main_supplies[i]; @@ -428,6 +448,13 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) goto err_enable; } switch (ret) { + case 0x1811: + devname = "WM1811"; + if (wm8994->type != WM1811) + dev_warn(wm8994->dev, "Device registered as type %d\n", + wm8994->type); + wm8994->type = WM1811; + break; case 0x8994: devname = "WM8994"; if (wm8994->type != WM8994) @@ -592,6 +619,7 @@ static int wm8994_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id wm8994_i2c_id[] = { + { "wm1811", WM1811 }, { "wm8994", WM8994 }, { "wm8958", WM8958 }, { } diff --git a/include/linux/mfd/wm8994/core.h b/include/linux/mfd/wm8994/core.h index a275798..f44bdb7 100644 --- a/include/linux/mfd/wm8994/core.h +++ b/include/linux/mfd/wm8994/core.h @@ -20,6 +20,7 @@ enum wm8994_type { WM8994 = 0, WM8958 = 1, + WM1811 = 2, }; struct regulator_dev; -- 1.7.6.3

1 2

[alsa-devel] [PATCH] ASoC: Add device ID for WM9093 to WM9090 driver
by Mark Brown 27 Sep '11

27 Sep '11

The WM9093 is an enhanced version of the WM9093. Add the device ID to the driver, further patches will add support for the additional features in the WM9093. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9090.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4de1220..6112ddf 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -684,6 +684,7 @@ static int __devexit wm9090_i2c_remove(struct i2c_client *i2c) static const struct i2c_device_id wm9090_id[] = { { "wm9090", 0 }, + { "wm9093", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, wm9090_id); -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ASoC: Add DRC control for WM8996
by Mark Brown 27 Sep '11

27 Sep '11

From: Karl Tsou <karl(a)opensource.wolfsonmicro.com> Signed-off-by: Karl Tsou <karl(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 833df74..b98a8f8 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -641,6 +641,14 @@ SOC_DOUBLE_R("Speaker ZC Switch", WM8996_LEFT_PDM_SPEAKER, SOC_SINGLE("DSP1 EQ Switch", WM8996_DSP1_RX_EQ_GAINS_1, 0, 1, 0), SOC_SINGLE("DSP2 EQ Switch", WM8996_DSP2_RX_EQ_GAINS_1, 0, 1, 0), + +SOC_SINGLE("DSP1 DRC TXL Switch", WM8996_DSP1_DRC_1, 0, 1, 0), +SOC_SINGLE("DSP1 DRC TXR Switch", WM8996_DSP1_DRC_1, 1, 1, 0), +SOC_SINGLE("DSP1 DRC RX Switch", WM8996_DSP1_DRC_1, 2, 1, 0), + +SOC_SINGLE("DSP2 DRC TXL Switch", WM8996_DSP2_DRC_1, 0, 1, 0), +SOC_SINGLE("DSP2 DRC TXR Switch", WM8996_DSP2_DRC_1, 1, 1, 0), +SOC_SINGLE("DSP2 DRC RX Switch", WM8996_DSP2_DRC_1, 2, 1, 0), }; static const struct snd_kcontrol_new wm8996_eq_controls[] = { -- 1.7.6.3

1 0

[alsa-devel] davinci-pcm debug
by David Pettit 27 Sep '11

27 Sep '11

I've written a new codec driver and tested it on an EVM for a TI Davinci ARM processor. It appeared to be working there. However, now I'm trying to debug buffer under-runs in my embedded system. I've inserted a debugging trace in the function davinci_pcm_pointer() in davinci-pdm.c to print out the value of asp_count. What I'm seeing is that when I first start playing an audio file, the value of asp_count increments very slowly. After some time, the value of asp_count start incrementing by the period size and I start seeing the debugging trace "davinci-pcm: audio_set_dma_params_play ...". Is this behavior expected? If I enable XRUN DEBUG and set the value to 5, then I occasionally see the message "PCM: hw_ptr skipping! (pos=17153, delta=23938, period=512, jdelta=0/49/0, hw_ptr=17279/17279)" where delta is always high. The comments in pcm_lib.c indicate that delta should be small or zero. What does a large value indicate? Thanks, David

2 2

Re: [alsa-devel] [PATCH v2] ALSA: hdmi: expose ELD control
by Takashi Iwai 27 Sep '11

27 Sep '11

At Tue, 27 Sep 2011 10:48:39 -0500, Pierre-Louis Bossart wrote: > > Thanks for your review Takashi! > > > Wouldn't it be better to strip the buffer data and size in the error > > case? For example, radeon driver gives often size=0. Then it's > > translated as size=128 for a workaround, then again handled as an > > error by checking the version. > > Yes, I wasn't sure what to do here and expected some comments. Should the > strategy be that if there's a single error we discard the entire ELD by > setting the size to zero? > No issues for the other changes. It's a good question. I'm inclined to discard all for any errors, in the case of ELD. AFAIK, most of errors are really broken ELD, and no need to care the rest of bits. > > Set the access field to SNDRV_CTL_ELEM_ACCESS_READ. > > Also, unless we implement a notification, better to set > > SNDRV_CTL_ELEM_ACCESS_VOLATILE flag, too. > > The intent was to read the ELD after the device is opened, I don't think the > ELD can change during playback, the updates can only happen on a pin-sensing > event. I can't think of a good reason why a notification would be needed? It's just a matter of definition. If the data may be changed without notification, the element should set VOLATILE flag. You don't have to implement the notification mechanism if you don't want :) thanks, Takashi

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[alsa-devel] [PATCH 1/3] ASoC: Allow DAI formats to be specified in the dai_link
by Mark Brown 27 Sep '11

27 Sep '11

For almost all machines the DAI format is a constant, always set to the same thing. This means that not only should we normally set it on init rather than in hw_params() (where it has been for historical reasons) we should also allow users to configure this by setting a variable in the dai_link structure. The combination of these two will make many machine drivers even more data driven. Implement a new dai_fmt field in the dai_link doing just that. Since 0 is a valid value for many format flags and we need to be able to tell if the field is actually set also add one to all the values used to configure formats. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 34 +++++++++++++++++----------------- include/sound/soc.h | 2 ++ sound/soc/soc-core.c | 22 ++++++++++++++++++++++ 3 files changed, 41 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 12d98b4..2413acc 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -24,13 +24,13 @@ struct snd_pcm_substream; * Describes the physical PCM data formating and clocking. Add new formats * to the end. */ -#define SND_SOC_DAIFMT_I2S 0 /* I2S mode */ -#define SND_SOC_DAIFMT_RIGHT_J 1 /* Right Justified mode */ -#define SND_SOC_DAIFMT_LEFT_J 2 /* Left Justified mode */ -#define SND_SOC_DAIFMT_DSP_A 3 /* L data MSB after FRM LRC */ -#define SND_SOC_DAIFMT_DSP_B 4 /* L data MSB during FRM LRC */ -#define SND_SOC_DAIFMT_AC97 5 /* AC97 */ -#define SND_SOC_DAIFMT_PDM 6 /* Pulse density modulation */ +#define SND_SOC_DAIFMT_I2S 1 /* I2S mode */ +#define SND_SOC_DAIFMT_RIGHT_J 2 /* Right Justified mode */ +#define SND_SOC_DAIFMT_LEFT_J 3 /* Left Justified mode */ +#define SND_SOC_DAIFMT_DSP_A 4 /* L data MSB after FRM LRC */ +#define SND_SOC_DAIFMT_DSP_B 5 /* L data MSB during FRM LRC */ +#define SND_SOC_DAIFMT_AC97 6 /* AC97 */ +#define SND_SOC_DAIFMT_PDM 7 /* Pulse density modulation */ /* left and right justified also known as MSB and LSB respectively */ #define SND_SOC_DAIFMT_MSB SND_SOC_DAIFMT_LEFT_J @@ -42,8 +42,8 @@ struct snd_pcm_substream; * DAI bit clocks can be be gated (disabled) when the DAI is not * sending or receiving PCM data in a frame. This can be used to save power. */ -#define SND_SOC_DAIFMT_CONT (0 << 4) /* continuous clock */ -#define SND_SOC_DAIFMT_GATED (1 << 4) /* clock is gated */ +#define SND_SOC_DAIFMT_CONT (1 << 4) /* continuous clock */ +#define SND_SOC_DAIFMT_GATED (2 << 4) /* clock is gated */ /* * DAI hardware signal inversions. @@ -51,10 +51,10 @@ struct snd_pcm_substream; * Specifies whether the DAI can also support inverted clocks for the specified * format. */ -#define SND_SOC_DAIFMT_NB_NF (0 << 8) /* normal bit clock + frame */ -#define SND_SOC_DAIFMT_NB_IF (1 << 8) /* normal BCLK + inv FRM */ -#define SND_SOC_DAIFMT_IB_NF (2 << 8) /* invert BCLK + nor FRM */ -#define SND_SOC_DAIFMT_IB_IF (3 << 8) /* invert BCLK + FRM */ +#define SND_SOC_DAIFMT_NB_NF (1 << 8) /* normal bit clock + frame */ +#define SND_SOC_DAIFMT_NB_IF (2 << 8) /* normal BCLK + inv FRM */ +#define SND_SOC_DAIFMT_IB_NF (3 << 8) /* invert BCLK + nor FRM */ +#define SND_SOC_DAIFMT_IB_IF (4 << 8) /* invert BCLK + FRM */ /* * DAI hardware clock masters. @@ -63,10 +63,10 @@ struct snd_pcm_substream; * i.e. if the codec is clk and FRM master then the interface is * clk and frame slave. */ -#define SND_SOC_DAIFMT_CBM_CFM (0 << 12) /* codec clk & FRM master */ -#define SND_SOC_DAIFMT_CBS_CFM (1 << 12) /* codec clk slave & FRM master */ -#define SND_SOC_DAIFMT_CBM_CFS (2 << 12) /* codec clk master & frame slave */ -#define SND_SOC_DAIFMT_CBS_CFS (3 << 12) /* codec clk & FRM slave */ +#define SND_SOC_DAIFMT_CBM_CFM (1 << 12) /* codec clk & FRM master */ +#define SND_SOC_DAIFMT_CBS_CFM (2 << 12) /* codec clk slave & FRM master */ +#define SND_SOC_DAIFMT_CBM_CFS (3 << 12) /* codec clk master & frame slave */ +#define SND_SOC_DAIFMT_CBS_CFS (4 << 12) /* codec clk & FRM slave */ #define SND_SOC_DAIFMT_FORMAT_MASK 0x000f #define SND_SOC_DAIFMT_CLOCK_MASK 0x00f0 diff --git a/include/sound/soc.h b/include/sound/soc.h index b499b37..a4dc699 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -713,6 +713,8 @@ struct snd_soc_dai_link { const char *cpu_dai_name; const char *codec_dai_name; + unsigned int dai_fmt; /* format to set on init */ + /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index bd20154..b0cec3c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1317,6 +1317,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; + struct snd_soc_dai_link *dai_link; int ret, i, order; mutex_lock(&card->mutex); @@ -1429,6 +1430,27 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); + for (i = 0; i < card->num_links; i++) { + dai_link = &card->dai_link[i]; + + if (dai_link->dai_fmt) { + pr_crit("%d %p\n", i, card->rtd[i].codec_dai); + ret = snd_soc_dai_set_fmt(card->rtd[i].codec_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].codec_dai->dev, + "Failed to set DAI format: %d\n", + ret); + + ret = snd_soc_dai_set_fmt(card->rtd[i].cpu_dai, + dai_link->dai_fmt); + if (ret != 0) + dev_warn(card->rtd[i].cpu_dai->dev, + "Failed to set DAI format: %d\n", + ret); + } + } + snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), -- 1.7.6.3

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[alsa-devel] [PATCH v2] ALSA: hdmi: expose ELD control
by Pierre-Louis Bossart 27 Sep '11

27 Sep '11

Applications may want to read ELD information to understand what codecs are supported on the HDMI receiver and handle the a-v delay for better lip-sync. ELD information is exposed in a device-specific IFACE_PCM kcontrol. Tested both with amixer and PulseAudio. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 9 ++-- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_hdmi.c | 88 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 94 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index c34f730..9bf0d8b 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -327,14 +327,14 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); size = 128; } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE || size > PAGE_SIZE) { snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); return -ERANGE; } - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; + /* update info */ + eld->eld_size = size; + buf = eld->eld_buffer; for (i = 0; i < size; i++) { unsigned int val = hdmi_get_eld_data(codec, nid, i); @@ -356,7 +356,6 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, ret = hdmi_update_eld(eld, buf, size); error: - kfree(buf); return ret; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index aaefa7c..04d730f 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -621,6 +621,7 @@ struct cea_sad { }; #define ELD_FIXED_BYTES 20 +#define ELD_MAX_SIZE 256 #define ELD_MAX_MNL 16 #define ELD_MAX_SAD 16 @@ -645,6 +646,7 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3f1f6ac..944e0cd 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -324,6 +324,86 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) return -EINVAL; } +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + + spec = codec->spec; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + + pin_idx = kcontrol->private_value; + snd_BUG_ON(pin_idx < 0); + + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + uinfo->count = eld->eld_size; + + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + int size; + + spec = codec->spec; + pin_idx = kcontrol->private_value; + + snd_BUG_ON(pin_idx < 0); + + memset(ucontrol->value.bytes.data, 0, ELD_MAX_SIZE); + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + size = eld->eld_size; + + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, size); + + /* reset rest of control */ + memset(&ucontrol->value.bytes.data[size], 0, ELD_MAX_SIZE-size); + + return 0; +} + +static struct snd_kcontrol_new eld_bytes_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, +}; + +static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, + int device) +{ + struct snd_kcontrol *kctl; + struct hdmi_spec *spec = codec->spec; + int err; + + if (pin_idx < 0) + return -EINVAL; + + kctl = snd_ctl_new1(&eld_bytes_ctl, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = pin_idx; + kctl->id.device = device; + + err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl); + if (err < 0) + return err; + + return 0; +} + #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) @@ -1193,6 +1273,14 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pin_idx); + + /* add control for ELD Bytes */ + err = hdmi_create_eld_ctl(codec, + pin_idx, + spec->pcm_rec[pin_idx].device); + + if (err < 0) + return err; } return 0; -- 1.7.6.2

2 2

[alsa-devel] ASoC updates for 3.2
by Mark Brown 27 Sep '11

27 Sep '11

The following changes since commit 0a742681e6072a71f30cfe6312f758f1cd185c21: ASoC: Add missed free_irq in wm5100_remove and wm5100_probe error path (2011-09-23 11:54:24 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Axel Lin (4): ASoC: Drop exporting sn95031_get_mic_bias ASoC: Drop exporting ad1980_dai ASoC: Remove unused fields in struct mfld_mc_private ASoC: Staticise sst_platform_dai Lars-Peter Clausen (3): ASoC: ssm2602: Re-enable oscillator after suspend ASoC: ssm2602: Set initial bias level to standby ASoC: ssm2602: Support setting the oscillator and the clock output state Mark Brown (9): ASoC: Support a wider range of sample rates on Speyside WM8962 ASoC: Rename WM8962 DMIC widget to DMIC_ENA ASoC: Convert WM8962 MICBIAS to a supply widget ASoC: Add support for on-board analogue microphones on Speyside WM8962 ASoC: Add DMIC control to Speyside WM8962 board ASoC: Don't force bias on ground referenced devices ASoC: Add platform data for WM1250 EV1 GPIOs ASoC: Set idle_bias_off for WM1250 EV1 Merge branch 'for-3.1' into for-3.2 Peter Ujfalusi (11): ASoC: omap-mcbsp: Fix compile time warning about ambiguous ‘else’ ASoC: twl6040: Read the TRIM values from the chip ASoC: twl6040: Function to fetch the TRIM values ASoC: omap-mcpdm: API to configure offset cancellation ASoC: sdp4430: Configure McPDM offset cancellation ASoC: twl6040: Rename pga_event to out_drv_event ASoC: twl6040: Combine the custom volsw get, and put functions ASoC: twl6040: Move delayed_work struct inside twl6040_output for HS/HF ASoC: twl6040: Move the delayed_work for HS detection under twl6040_jack_data ASoC: twl6040: No need to change delay during HS ramp ASoC: twl6040: No need to change delay during HF ramp include/sound/wm1250-ev1.h | 27 ++++++ sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/sn95031.c | 1 - sound/soc/codecs/ssm2602.c | 73 +++++++++++++---- sound/soc/codecs/ssm2602.h | 6 +- sound/soc/codecs/twl6040.c | 155 ++++++++++++----------------------- sound/soc/codecs/twl6040.h | 13 +++ sound/soc/codecs/wm1250-ev1.c | 122 +++++++++++++++++++++++++++- sound/soc/codecs/wm8962.c | 10 +- sound/soc/mid-x86/mfld_machine.c | 2 - sound/soc/mid-x86/sst_platform.c | 2 +- sound/soc/omap/omap-mcbsp.c | 3 +- sound/soc/omap/omap-mcpdm.c | 25 ++++++ sound/soc/omap/omap-mcpdm.h | 12 +++ sound/soc/omap/sdp4430.c | 10 ++- sound/soc/samsung/speyside_wm8962.c | 24 ++++-- sound/soc/soc-dapm.c | 7 +- 17 files changed, 350 insertions(+), 143 deletions(-) create mode 100644 include/sound/wm1250-ev1.h

2 1

[alsa-devel] [PATCH 00/10] ASoC: twl6040: Gain ramp code cleanups
by Peter Ujfalusi 27 Sep '11

27 Sep '11

Hello, 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 (10): ASoC: twl6040: Rename pga_event to out_drv_event ASoC: twl6040: Combine the custom volsw get, and put functions ASoC: twl6040: Move delayed_work struct inside twl6040_output for HS/HF ASoC: twl6040: Move the delayed_work for HS detection under twl6040_jack_data ASoC: twl6040: One workqueue should be enough ASoC: twl6040: correct loop counters for HS/HF ramp code ASoC: twl6040: No need to change delay during HS ramp ASoC: twl6040: No need to change delay during HF ramp 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 | 203 ++++++++++++------------------------------- 1 files changed, 57 insertions(+), 146 deletions(-) -- 1.7.6.1

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[alsa-devel] [PATCH 1/3] ASoC: ssm2602: Re-enable oscillator after suspend
by Lars-Peter Clausen 27 Sep '11

27 Sep '11

Currently the the internal oscillator is powered down when entering BIAS_OFF state, but not re-enabled when going back to BIAS_STANDBY. As a result the CODEC will stop working after suspend if the internal oscillator is used to generate the sysclock signal. This patch fixes it by clearing the appropriate bit in the power down register when the CODEC is re-enabled. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Cc: stable(a)kernel.org --- sound/soc/codecs/ssm2602.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index cceb002..32d6c51 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -430,7 +430,8 @@ static int ssm2602_set_dai_fmt(struct snd_soc_dai *codec_dai, static int ssm2602_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, SSM2602_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, SSM2602_PWR); + reg &= ~(PWR_POWER_OFF | PWR_OSC_PDN); switch (level) { case SND_SOC_BIAS_ON: -- 1.7.2.5

2 6

[alsa-devel] ASoC updates for 3.1
by Mark Brown 27 Sep '11

27 Sep '11

The following changes since commit f0e8ed858edb327802ee65fd695cc1538286226f: ASoC: Ensure we generate a driver name (2011-09-21 14:54:23 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.1 Jarkko Nikula (1): ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active Lars-Peter Clausen (1): ASoC: ssm2602: Re-enable oscillator after suspend sound/soc/codecs/ssm2602.c | 3 ++- sound/soc/omap/omap-mcbsp.c | 6 ++++++ 2 files changed, 8 insertions(+), 1 deletions(-)

2 1

[alsa-devel] [PATCH 1/2] ASoC: Remove unused fields in struct mfld_mc_private
by Axel Lin 27 Sep '11

27 Sep '11

Both *socdev and *codec of struct mfld_mc_private are not being used in this driver, remove it. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/mid-x86/mfld_machine.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/mid-x86/mfld_machine.c b/sound/soc/mid-x86/mfld_machine.c index 429aa1b..1d818dc 100644 --- a/sound/soc/mid-x86/mfld_machine.c +++ b/sound/soc/mid-x86/mfld_machine.c @@ -54,9 +54,7 @@ static unsigned int hs_switch; static unsigned int lo_dac; struct mfld_mc_private { - struct platform_device *socdev; void __iomem *int_base; - struct snd_soc_codec *codec; u8 interrupt_status; }; -- 1.7.4.1

2 3

[alsa-devel] [RFC][PATCH] ASoC: Make wm1250-ev1 depends on GPIOLIB
by Axel Lin 27 Sep '11

27 Sep '11

I got below build error if GPIO support is disabled. Make wm1250-ev1 depends on GPIOLIB to fix below build error. CC sound/soc/codecs/wm1250-ev1.o sound/soc/codecs/wm1250-ev1.c:32:14: error: array type has incomplete element type sound/soc/codecs/wm1250-ev1.c: In function ‘wm1250_ev1_pdata’: sound/soc/codecs/wm1250-ev1.c:126:18: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:126:18: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:126:18: error: negative width in bit-field ‘<anonymous>’ sound/soc/codecs/wm1250-ev1.c:134:42: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:134:42: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:134:42: error: negative width in bit-field ‘<anonymous>’ sound/soc/codecs/wm1250-ev1.c: In function ‘wm1250_ev1_free’: sound/soc/codecs/wm1250-ev1.c:155:34: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:155:34: warning: type defaults to ‘int’ in type name sound/soc/codecs/wm1250-ev1.c:155:34: error: negative width in bit-field ‘<anonymous>’ make[3]: *** [sound/soc/codecs/wm1250-ev1.o] Error 1 make[2]: *** [sound/soc/codecs] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- I got the build error while trying to build for SND_MFLD_MACHINE and SND_SST_PLATFORM. ( "Enable ALSA for SoC audio support" and "Build all ASoC CODEC drivers" on X86 ) Axel sound/soc/codecs/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 3449431..0ad80fe 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -58,7 +58,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE - select SND_SOC_WM1250_EV1 if I2C + select SND_SOC_WM1250_EV1 if I2C && GPIOLIB select SND_SOC_WM2000 if I2C select SND_SOC_WM5100 if I2C select SND_SOC_WM8350 if MFD_WM8350 -- 1.7.4.1

2 1

[alsa-devel] [patch] ALSA: usb-audio - NULL dereference on probe error
by Dan Carpenter 27 Sep '11

27 Sep '11

There was a potential NULL dereference introduced in 362e4e49abe "ALSA: usb-audio - clear chip->probing on error exit" Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/usb/card.c b/sound/usb/card.c index cc08588..8db260f 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -532,7 +532,8 @@ snd_usb_audio_probe(struct usb_device *dev, __error: if (chip && !chip->num_interfaces) snd_card_free(chip->card); - chip->probing = 0; + if (chip) + chip->probing = 0; mutex_unlock(&register_mutex); __err_val: return NULL;

2 1

[alsa-devel] [PATCH] Set CLOEXEC flag for pipes in PulseAudio plugins
by Chris Rankin 27 Sep '11

27 Sep '11

Hi, I have noticed that Alsa / PulseAudio are leaking a couple of pipes after a fork/exec, and have traced the leak to the following two plugins: libasound_module_ctl_pulse.so libasound_module_pcm_pulse.so This is with Fedora 15. I have attached a patch for the problem. Cheers, Chris

2 1

Re: [alsa-devel] [PATCH] sound: usb: mixer: increase control transfer timeout
by Takashi Iwai 27 Sep '11

27 Sep '11

At Mon, 29 Aug 2011 14:10:26 +0300, Felipe Balbi wrote: > > 100ms might not be enough for slower devices. Increasing > timeout to 1000ms doesn't seem like overkill. > > Signed-off-by: Felipe Balbi <balbi(a)ti.com> Felipe, what is the status of this patch? Could you resend a revised patch? thanks, Takashi

3 2

Re: [alsa-devel] snd_pcm_pause kills plaback handle on OMAP4
by john＠johnandjuliet.com 27 Sep '11

27 Sep '11

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

2 1

[alsa-devel] "modprobe snd_hda_intel" crashes on my hardware
by Abdelghani Ouchabane 27 Sep '11

27 Sep '11

Description of problem: I have a customized sound hardware, it looks that it doesn't follow the standard 100 %. I had to patch the sound driver under fedora 9 ( 2.6.27.25-78.2.56.fc9.i686 ), exactly I patched ./sound/pci/hda/patch_via.c by forcing the driver to program the codec. It was working fine in Fedora 9. But in Fedora 15, I have tried the patch the ./sound/pci/hda/patch_via.c, but unfortunately the path doesn't work. Please : could you help me to figure out how to patch ./sound/pci/hda/patch_via.c to make sound works in my hardware. Version-Release number of selected component (if applicable): kernel-2.6.40.3-0.fc15.i686.rpm Additional info: **************************************************************************************************************************************************************************** dmesg log : [ 95.366164] HDA Intel 0000:00:1b.0: PCI INT A -> Link[LNKA] -> GSI 10 (level, low) -> IRQ 10 [ 95.366209] HDA Intel 0000:00:1b.0: setting latency timer to 64 [ 95.398391] hda_codec: Cannot set up configuration from BIOS. Using genenic mode... [ 95.398470] BUG: unable to handle kernel NULL pointer dereference at (null) [ 95.398477] IP: [<f83740ac>] via_build_pcms+0x60/0xdc [snd_hda_codec_via] [ 95.398494] *pde = 3173f067 *pte = 00000000 [ 95.398501] Oops: 0000 [#1] PREEMPT [ 95.398509] Modules linked in: snd_hda_codec_via snd_hda_intel(+) snd_hda_codec snd_hwdep snd_seq snd_seq_device snd_pcm snd_timer snd soundcore snd_page_alloc ipv6 fakephp cgosdrv(P) joydev mcs7830 usbnet mii usbtouchscreen iTCO_wdt i2c_i801 pciehp iTCO_vendor_support serio_raw xfs i915 drm_kms_helper drm i2c_algo_bit i2c_core video [last unloaded: scsi_wait_scan] [ 95.398558] [ 95.398564] Pid: 4208, comm: modprobe Tainted: P 2.6.40.3-0.119.delos.i686 #1 [ 95.398572] EIP: 0060:[<f83740ac>] EFLAGS: 00010246 CPU: 0 [ 95.398581] EIP is at via_build_pcms+0x60/0xdc [snd_hda_codec_via] [ 95.398587] EAX: f174f000 EBX: 00000000 ECX: 00000000 EDX: dd58f400 [ 95.398592] ESI: f837c760 EDI: f174f120 EBP: f17afd28 ESP: f17afd20 [ 95.398597] DS: 007b ES: 007b FS: 0000 GS: 00e0 SS: 0068 [ 95.398603] Process modprobe (pid: 4208, ti=f17ae000 task=f0e5be30 task.ti=f17ae000) [ 95.398607] Stack: [ 95.398610] dd58f400 00000001 f17afd68 f81ef28a 00000050 f81f6511 f16aa960 f16aa968 [ 95.398622] 00000001 11061708 2d646e73 2d616468 65646f63 64692d63 3031313a dd58f400 [ 95.398634] f578642c 00000001 f17afd78 f81f5aaf f171da00 f5786424 f17afde4 f816e6fd [ 95.398645] Call Trace: [ 95.398665] [<f81ef28a>] snd_hda_codec_build_pcms+0x25/0x230 [snd_hda_codec] [ 95.398685] [<f81f5aaf>] snd_hda_build_pcms+0x17/0x2e [snd_hda_codec] [ 95.398699] [<f816e6fd>] azx_probe+0x9a6/0xa44 [snd_hda_intel] [ 95.398713] [<f816d1cd>] ? azx_pcm_open+0x1c8/0x1c8 [snd_hda_intel] [ 95.398725] [<f816d394>] ? azx_acquire_irq+0x9b/0x9b [snd_hda_intel] [ 95.398737] [<f816dbb9>] ? kzalloc.constprop.5+0x14/0x14 [snd_hda_intel] [ 95.398749] [<f816db45>] ? azx_resume+0xfa/0xfa [snd_hda_intel] [ 95.398761] [<f816d9e7>] ? azx_init_chip+0x1d5/0x1d5 [snd_hda_intel] [ 95.398775] [<c05dfcc1>] pci_device_probe+0x5e/0xa7 [ 95.398787] [<c06616ef>] driver_probe_device+0x121/0x1fc [ 95.398794] [<c05dfc39>] ? pci_match_device+0x5d/0x68 [ 95.398802] [<c0661812>] __driver_attach+0x48/0x64 [ 95.398810] [<c0660931>] bus_for_each_dev+0x42/0x6b [ 95.398817] [<c0661350>] driver_attach+0x1f/0x23 [ 95.398824] [<c06617ca>] ? driver_probe_device+0x1fc/0x1fc [ 95.398832] [<c0660fcb>] bus_add_driver+0xca/0x210 [ 95.398840] [<c0661c2a>] driver_register+0x84/0xe3 [ 95.398848] [<c05e03f7>] __pci_register_driver+0x34/0x91 [ 95.398862] [<f820c017>] alsa_card_azx_init+0x17/0x19 [snd_hda_intel] [ 95.398871] [<c0401173>] do_one_initcall+0x8b/0x138 [ 95.398878] [<f820c000>] ? 0xf820bfff [ 95.398889] [<c045945f>] sys_init_module+0x12bd/0x14dc [ 95.398905] [<c07c0e94>] syscall_call+0x7/0xb [ 95.398909] Code: 00 f3 a5 8d b8 f0 00 00 00 8b 48 50 8b 70 3c 0f b7 09 66 89 88 cc 00 00 00 b9 0c 00 00 00 f3 a5 8b 88 9c 00 00 00 66 83 78 5c 00 <0f> b7 09 66 89 88 fc 00 00 00 8b 48 64 89 88 c8 00 00 00 75 0a [ 95.398972] EIP: [<f83740ac>] via_build_pcms+0x60/0xdc [snd_hda_codec_via] SS:ESP 0068:f17afd20 [ 95.398984] CR2: 0000000000000000 [ 95.488158] ---[ end trace e7a4603bcd7d8946 ]--- **************************************************************************************************************************************************************************** lspci -v : 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) Flags: bus master, fast devsel, latency 0, IRQ 10 Memory at fea38000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable- Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel Capabilities: [130] Root Complex Link Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel **************************************************************************************************************************************************************************** * Please, I need your great help to make my sound card works as I did for Fedora 9. * Or any workaround. * If you need any information, please don't hesitate. **************************************************************************************************************************************************************************** * I have used the attached patch to force the driver to program the code. **************************************************************************************************************************************************************************** Thanks a lot in advance.

1 1

Re: [alsa-devel] [PATCH] ASoC: fsl: Fix error handling if platform_device_add fails
by Tabi Timur-B04825 27 Sep '11

27 Sep '11

Axel Lin wrote: > Call platform_device_put() instead of platform_device_unregister() if > platform_device_add() fails. What's wrong with calling platform_device_unregister()? -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] [PATCH] ASoC: Flush Samsung DMA on free
by Mark Brown 27 Sep '11

27 Sep '11

Ever since it was written the Samsung DMA driver has had a TODO in the hw_free() function wondering if we need to flush the DMA buffers. Up until now the answer has been no but with the recent improvements Boojin has done to the DMA infrastructure for the Samsung port the answer has changed to yes for at least S3C6410 systems. If we don't then when we next prepare() the channel the API will get confused trying to run callbacks on the transfers hanging around from the previous time the stream was open and oops. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Kukjin, this should probably get added to your branch assuming it's OK for other CPUs. sound/soc/samsung/dma.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 851346f..2d622b6 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -198,10 +198,10 @@ static int dma_hw_free(struct snd_pcm_substream *substream) pr_debug("Entered %s\n", __func__); - /* TODO - do we need to ensure DMA flushed */ snd_pcm_set_runtime_buffer(substream, NULL); if (prtd->params) { + prtd->params->ops->flush(prtd->params->ch); prtd->params->ops->release(prtd->params->ch, prtd->params->client); prtd->params = NULL; -- 1.7.6.3

2 1

Re: [alsa-devel] Questions for dummy codec driver
by Mark Brown 27 Sep '11

27 Sep '11

On Thu, Sep 22, 2011 at 09:42:14AM +0800, Leo Yan wrote: > On our platform, there have some devices (like hdmi/bt) do not need > config from the CODEC driver, they have their own s/w stack and > interface to configure; for the audio driver, just need to set the > I2S/PCM timing for them. So if there have a common dummy codec > driver, then it can meet our requirement well. Otherwise, there will > have some duplicate codes for these dummy codecs. The general thought here has been that for dumb CODECs the boilerplate is sufficiently small and simple to mean that the discoverability from writing a skeletal driver outweighs the cost of having it kicking around - while there is some boilerplate there it's easier for someone to grab and use this code than it is to work out how exactly to translate the datasheets into code each time they encounter the part. However, a sufficiently well written generic driver would maintain that advantage while reducing the duplication.

1 0

Re: [alsa-devel] Adding ELD control to alsa-lib hdmi device
by David Henningsson 26 Sep '11

26 Sep '11

On 09/23/2011 08:46 PM, Pierre-Louis Bossart wrote: > Hi, > Now that I almost done with exposing a read-only ELD control for HDA/HDMI > outputs, I was wondering how to read it from user-land. PulseAudio and > others rely on the 'hdmi' device, and apparently I can't access this new ELD > control, I have to use the 'hw:x,y' details. The hdmi:x(,y) device is only used for pcm. For opening the mixer, hw:x (never hw:x,y - that's pcm only as well) is used as the device string. > I really don't understand much to alsa-lib... Can I expose this new control > through the ALSA configuration files where the hdmi device is defined? Or is > there a way to infer the card and device number actually used by an alsa-lib > plugin? Or do I have to add code to propagate the information from > hardware-level to higher-level plugins? > Thanks for your feedback! I'm not the expert here, but if I were you, I would run amixer -D hw:x contents amixer -D hw:x scontents To see if your controls show up there. If it shows up in scontents, that would be the "simple" mixer interface that PulseAudio uses. If not, using the control api (as being done by "contents") should work. Anyway, reading the source code to amixer would then be the first step towards understanding how to enumerate and read mixer controls. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH] usb-audio: Check for possible chip NULL pointer before clearing probing flag
by Thomas Pfaff 26 Sep '11

26 Sep '11

Before clearing the probing flag in the error exit path, check that the chip pointer is not NULL. Signed-off-by: Thomas Pfaff <tpfaff(a)gmx.net> --- diff -urp a/sound/usb/card.c b/sound/usb/card.c --- a/sound/usb/card.c 2011-09-26 14:58:02.468672118 +0200 +++ b/sound/usb/card.c 2011-09-26 15:00:30.313110086 +0200 @@ -530,9 +530,11 @@ snd_usb_audio_probe(struct usb_device *d return chip; __error: - if (chip && !chip->num_interfaces) - snd_card_free(chip->card); - chip->probing = 0; + if (chip) { + if (!chip->num_interfaces) + snd_card_free(chip->card); + chip->probing = 0; + } mutex_unlock(&register_mutex); __err_val: return NULL;

2 1

[alsa-devel] [PATCH resend 0/4] ASoC: omap-mcpdm/twl6040: Offset cancellation
by Peter Ujfalusi 26 Sep '11

26 Sep '11

Hello, The following series adds support for the McPDM offset cancellation feature. It provides the needed functionality to: - read the HSTRIM values from the twl6040 codec - configure the offset cancellation registers in OMAP-McPDM Rebased batches on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc, for-3.2 branch No other changes. Mark: I have not added you acks to the patches, since you will be signing them off anyways. Regards, Peter --- Peter Ujfalusi (4): ASoC: twl6040: Read the TRIM values from the chip ASoC: twl6040: Function to fetch the TRIM values ASoC: omap-mcpdm: API to configure offset cancellation ASoC: sdp4430: Configure McPDM offset cancellation sound/soc/codecs/twl6040.c | 16 ++++++++++++++++ sound/soc/codecs/twl6040.h | 13 +++++++++++++ sound/soc/omap/omap-mcpdm.c | 25 +++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 12 ++++++++++++ sound/soc/omap/sdp4430.c | 10 +++++++++- 5 files changed, 75 insertions(+), 1 deletions(-) -- 1.7.6.1

1 4

[alsa-devel] [PATCH 0/4] ASoC: omap-mcpdm/twl6040: Offset cancellation
by Peter Ujfalusi 26 Sep '11

26 Sep '11

Hello, The following series adds support for the McPDM offset cancellation feature. It provides the needed functionality to: - read the HSTRIM values from the twl6040 codec - configure the offset cancellation registers in OMAP-McPDM Regards, Peter --- Peter Ujfalusi (4): ASoC: twl6040: Read the TRIM values from the chip ASoC: twl6040: Function to fetch the TRIM values ASoC: omap-mcpdm: API to configure offset cancellation ASoC: sdp4430: Configure McPDM offset cancellation sound/soc/codecs/twl6040.c | 16 ++++++++++++++++ sound/soc/codecs/twl6040.h | 13 +++++++++++++ sound/soc/omap/omap-mcpdm.c | 25 +++++++++++++++++++++++++ sound/soc/omap/omap-mcpdm.h | 12 ++++++++++++ sound/soc/omap/sdp4430.c | 10 +++++++++- 5 files changed, 75 insertions(+), 1 deletions(-) -- 1.7.6.1

2 9

[alsa-devel] ASoC updates for 3.1
by Mark Brown 26 Sep '11

26 Sep '11

The following changes since commit 5013951be88e136d9990ef55303276e2779ce8d8: ASoC: Fix trivial build regression in Kirkwood I2S (2011-09-12 11:48:12 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.1 A couple of driver fixes and the driver name issue we discussed. Axel Lin (1): ASoC: bf5xx-ad73311: Fix prototype for bf5xx_probe Mark Brown (2): ASoC: Remove bitrotted wm8962_resume() ASoC: Ensure we generate a driver name sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/codecs/wm8962.c | 26 -------------------------- sound/soc/soc-core.c | 18 +++++++++++++++--- 3 files changed, 16 insertions(+), 30 deletions(-)

3 3

Re: [alsa-devel] link between HDMI ELD and PCM devices
by David Henningsson 26 Sep '11

26 Sep '11

On 09/21/2011 03:20 PM, Pierre-Louis Bossart wrote: >> This won't work - in the case of several HDMI codecs (common with >> NVidia) there will be more than one "HDMI 0". > > No, the code doesn't use a constant, It's actually four constants, please see generic_hdmi_pcm_names in patch_hdmi.c. But my point is that this restarts from 0 on every codec. > it's assigned in the same way as the > device name. You would have HDMI 0, HDMI 1..4. I don't have this kind of > hardware, it'd be good if someone could verify that the name is indeed > dynamic. On a 3.0 based kernel, here's part of my aplay -l output: card 3: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: NVidia [HDA NVidia], device 7: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: NVidia [HDA NVidia], device 8: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 3: NVidia [HDA NVidia], device 9: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 >> The solution is to actually find the device number out, like this: >> >> struct hdmi_spec *spec = codec->spec; >> int pcmdev = spec->pcm_rec[pin_idx].device; > > That's what I wanted initially, but this device field is not > used/initialized, and like I said the device assignment comes after the pin > sensing and all the ELD handling A good place to do this is generic_hdmi_build_controls, both because it is normally used to add controls (which is what you do), and because it is triggered after the pcm device is assigned. (A side note to Takashi: for some reason hda-emu has the calls to build_controls and build_pcms switched, compared to the real implementation.) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 6

[alsa-devel] [PATCH] ASoC: sdp4430: Fix string for FM input name
by Peter Ujfalusi 26 Sep '11

26 Sep '11

The name contains invalid valid character (/), which causes problems when trying to create the debugfs directory structure: ASoC: Failed to create Aux/FM Stereo In debugfs file Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/sdp4430.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index a951774..d462f1a 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -88,7 +88,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), + SND_SOC_DAPM_INPUT("FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -113,8 +113,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Earphone Spk", NULL, "EP"}, /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "Aux/FM Stereo In"}, - {"AFMR", NULL, "Aux/FM Stereo In"}, + {"AFML", NULL, "FM Stereo In"}, + {"AFMR", NULL, "FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) -- 1.7.6

3 4

Re: [alsa-devel] [PATCH] ASoC: Drop exporting ad1980_dai
by Mark Brown 26 Sep '11

26 Sep '11

On Sat, Sep 24, 2011 at 06:43:43PM +0800, Axel Lin wrote: > ad1980_dai is not used outside this driver, > thus drop exporting it. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: Drop exporting sn95031_get_mic_bias
by Mark Brown 26 Sep '11

26 Sep '11

On Sat, Sep 24, 2011 at 06:16:15PM +0800, Axel Lin wrote: > sn95031_get_mic_bias() is not used outside this driver > and it is a static function now. > Thus drop exporting sn95031_get_mic_bias. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Fix compile time warning about ambiguous ‘else’
by Peter Ujfalusi 26 Sep '11

26 Sep '11

Fixes the following compile time warning: omap-mcbsp.c:519: warning: suggest explicit braces to avoid ambiguous ‘else’ Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-mcbsp.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 478d607..1391ea0 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,11 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; - if (mcbsp_data->active) + if (mcbsp_data->active) { if (freq == mcbsp_data->in_freq) return 0; else return -EBUSY; + } /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || -- 1.7.6.1

3 2

[alsa-devel] [PATCH 0/8] ASoC/MFD/Input: twl6040: Support for audio driven vibra
by Peter Ujfalusi 26 Sep '11

26 Sep '11

Hello, The following series adds support for the soc driver to route audio (PCM stream) to the vibra ports of the twl6040. We have ALSA controls for the vibra source selection (PDM, or Input/FF). The route can not be changed from Input/FF to PDM during ongoing effect coming from Input/FF. The Input/FF driver can not execute new effect, while the vibra source is configured to be PDM. TO get things working in legacy mode the omap-mcpdm, and twl6040 soc codec driver's channel count needed to be fixed. To minimize the need for reads through the I2C bus, the MFD driver will cache the vibra control registers. With this series we can route PCM audio to drive the twl6040 vibra output. In order to that we need to play 5 channel audio, and place the vibra sample at the 5th channel. This will change with ABE support (change == simplified). The series depends on the previous two from me: ASoC: twl6040: Cleanups and fixes ASoC: omap-mcpdm/twl6040: Offset cancellation Regards, Peter PS: Samuel, Dmitry, Liam: will it be possible to take this through Liam (via audio), since we are going to have dependencies coming via audio. --- Peter Ujfalusi (8): Input: twl6040: Simplify vibra regsiter definitions MFD: twl6040: Cache the vibra control registers MFD: twl6040: function to query the vibra status for clients Input: twl6040-vibra: Check the selected path for vibra ASoC: twl6040: Fix the number of channels for vibra ASoC: twl6040: Correct supported number of playback channels ASoC: omap-mcpdm: Correct the supported number of channels ASoC: twl6040: Support for vibra output paths drivers/input/misc/twl6040-vibra.c | 19 ++++++--- drivers/mfd/twl6040-core.c | 31 ++++++++++++-- include/linux/mfd/twl6040.h | 24 +++++------ sound/soc/codecs/twl6040.c | 78 ++++++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcpdm.c | 14 ++++--- 5 files changed, 134 insertions(+), 32 deletions(-) -- 1.7.6.1

4 25

Re: [alsa-devel] Adding ELD control to alsa-lib hdmi device
by Raymond Yau 25 Sep '11

25 Sep '11

2011/9/24 Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: > Hi, > Now that I almost done with exposing a read-only ELD control for HDA/HDMI > outputs, I was wondering how to read it from user-land. PulseAudio and > others rely on the 'hdmi' device, and apparently I can't access this new ELD > control, I have to use the 'hw:x,y' details. If you are using iface_pcm http://www.alsa-project.org/alsa-doc/alsa-lib/control_8c.html CTL interface use snd_ctl_open() to open "hw:x" or use alsactl store

1 0

[alsa-devel] [PATCH 0/3] Fallback mechanism for pulse plugin
by Takashi Iwai 25 Sep '11

25 Sep '11

Hi, the following are experimental patches for implementing the fallback option of PCM / control pulse plugin. When the connection to PA server fails, the plugin tries to open the fallback name. For achieving this, I added the new standard definition "sysdefault", which is equal with the normal "default" PCM / control definitions. The difference is only the name, i.e. it won't be overridden by other setups. Then two new API functions for opening a fallback PCM / control, and finally a patch for pulse-plugin will follow. All changes are relatively small and easy. Let me know if you have any suggestions or a better idea. thanks, Takashi

5 23

Re: [alsa-devel] [PATCH] ALSA: snd-aloop: make preallocated buffer size configurable
by Takashi Iwai 24 Sep '11

24 Sep '11

At Wed, 21 Sep 2011 10:02:27 -0500, Pierre-Louis Bossart wrote: > > > I meant to have a kconfig option to limit the upper-bound of pre-alloc > > buffer sizes for all drivers. That is, a patch like below (untested) > > That's a good idea, but the two patches aren't necessarily exclusive. In an > embedded solution you may want a 2s PCM buffer for low-power playback and a > 64k buffer for snd-aloop as it is only used with small periods. Also one > would need to check the buffer/period sizes in each driver, and make sure > they are smaller than this global max. Well, if it's only about snd-aloop, another option would be to use vmalloc'ed buffer for aloop. Since it's no hardware buffer, it's more system-friendly, and you don't need a pre-allocation. Again, an untested patch is below. thanks, Takashi --- diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index a0da775..4067f15 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -575,7 +575,8 @@ static void loopback_runtime_free(struct snd_pcm_runtime *runtime) static int loopback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(params)); } static int loopback_hw_free(struct snd_pcm_substream *substream) @@ -587,7 +588,7 @@ static int loopback_hw_free(struct snd_pcm_substream *substream) mutex_lock(&dpcm->loopback->cable_lock); cable->valid &= ~(1 << substream->stream); mutex_unlock(&dpcm->loopback->cable_lock); - return snd_pcm_lib_free_pages(substream); + return snd_pcm_lib_free_vmalloc_buffer(substream); } static unsigned int get_cable_index(struct snd_pcm_substream *substream) @@ -740,6 +741,8 @@ static struct snd_pcm_ops loopback_playback_ops = { .prepare = loopback_prepare, .trigger = loopback_trigger, .pointer = loopback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops loopback_capture_ops = { @@ -751,6 +754,8 @@ static struct snd_pcm_ops loopback_capture_ops = { .prepare = loopback_prepare, .trigger = loopback_trigger, .pointer = loopback_pointer, + .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static int __devinit loopback_pcm_new(struct loopback *loopback, @@ -771,10 +776,6 @@ static int __devinit loopback_pcm_new(struct loopback *loopback, strcpy(pcm->name, "Loopback PCM"); loopback->pcm[device] = pcm; - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - snd_dma_continuous_data(GFP_KERNEL), - 0, 2 * 1024 * 1024); return 0; }

2 2

[alsa-devel] [PATCH] ALSA: HDA: No power nids on 92HD93
by David Henningsson 24 Sep '11

24 Sep '11

This patch is necessary to make internal speakers work on this chip. Cc: stable(a)kernel.org BugLink: http://bugs.launchpad.net/bugs/854468 Tested-by: Alex Wolfson <alex.wolfson(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1b7c114..987e3cf 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5630,6 +5630,7 @@ again: switch (codec->vendor_id) { case 0x111d76d1: case 0x111d76d9: + case 0x111d76df: case 0x111d76e5: case 0x111d7666: case 0x111d7667: -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Remove direct register cache accesses from WM8962 driver
by Mark Brown 23 Sep '11

23 Sep '11

Also fix return values for speaker switch updates. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 21 +++++++++++---------- 1 files changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 59b2ad6..c2f68aa 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2141,7 +2141,6 @@ static int wm8962_put_spk_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u16 *reg_cache = codec->reg_cache; int ret; /* Apply the update (if any) */ @@ -2150,16 +2149,19 @@ static int wm8962_put_spk_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_SPKOUTL_PGA_ENA) - return snd_soc_write(codec, WM8962_SPKOUTL_VOLUME, - reg_cache[WM8962_SPKOUTL_VOLUME]); + ret = snd_soc_read(codec, WM8962_PWR_MGMT_2); + if (ret & WM8962_SPKOUTL_PGA_ENA) { + snd_soc_write(codec, WM8962_SPKOUTL_VOLUME, + snd_soc_read(codec, WM8962_SPKOUTL_VOLUME)); + return 1; + } /* ...otherwise the right. The VU is stereo. */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_SPKOUTR_PGA_ENA) - return snd_soc_write(codec, WM8962_SPKOUTR_VOLUME, - reg_cache[WM8962_SPKOUTR_VOLUME]); + if (ret & WM8962_SPKOUTR_PGA_ENA) + snd_soc_write(codec, WM8962_SPKOUTR_VOLUME, + snd_soc_read(codec, WM8962_SPKOUTR_VOLUME)); - return 0; + return 1; } static const char *cap_hpf_mode_text[] = { @@ -2500,7 +2502,6 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - u16 *reg_cache = codec->reg_cache; int reg; switch (w->shift) { @@ -2523,7 +2524,7 @@ static int out_pga_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: - return snd_soc_write(codec, reg, reg_cache[reg]); + return snd_soc_write(codec, reg, snd_soc_read(codec, reg)); default: BUG(); return -EINVAL; -- 1.7.6.3

1 0

[alsa-devel] [PATCH 1/3] ASoC: Convert WM8962 MICBIAS to a supply widget
by Mark Brown 23 Sep '11

23 Sep '11

A supply widget is generally clearer than a MICBIAS widget and a mic bias is just a type of supply so use a supply widget for the MICBIAS. This also avoids confusion with the routing when connected to multiple inputs. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 2 +- sound/soc/samsung/speyside_wm8962.c | 10 +++++----- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 8679f06..59b2ad6 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2669,7 +2669,7 @@ SND_SOC_DAPM_INPUT("IN4R"), SND_SOC_DAPM_INPUT("Beep"), SND_SOC_DAPM_INPUT("DMICDAT"), -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8962_PWR_MGMT_1, 1, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS", WM8962_PWR_MGMT_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("Class G", WM8962_CHARGE_PUMP_B, 0, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("SYSCLK", WM8962_CLOCKING2, 5, 0, sysclk_event, diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 98b28ea..35539eb 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -152,12 +152,12 @@ static struct snd_soc_dapm_route audio_paths[] = { { "Main Speaker", NULL, "SPKOUTL" }, { "Main Speaker", NULL, "SPKOUTR" }, - { "MICBIAS", NULL, "Headset Mic" }, - { "IN4L", NULL, "MICBIAS" }, - { "IN4R", NULL, "MICBIAS" }, + { "Headset Mic", NULL, "MICBIAS" }, + { "IN4L", NULL, "Headset Mic" }, + { "IN4R", NULL, "Headset Mic" }, - { "MICBIAS", NULL, "DMIC" }, - { "DMICDAT", NULL, "MICBIAS" }, + { "DMIC", NULL, "MICBIAS" }, + { "DMICDAT", NULL, "DMIC" }, }; static struct snd_soc_jack speyside_wm8962_headset; -- 1.7.6.3

2 3

[alsa-devel] [PATCH] ASoC: Rename WM8962 DMIC widget to DMIC_ENA
by Mark Brown 23 Sep '11

23 Sep '11

Matches the register name and avoids confusion with board widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 431b976..59b2ad6 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2690,7 +2690,7 @@ SND_SOC_DAPM_MIXER("MIXINL", WM8962_PWR_MGMT_1, 5, 0, SND_SOC_DAPM_MIXER("MIXINR", WM8962_PWR_MGMT_1, 4, 0, mixinr, ARRAY_SIZE(mixinr)), -SND_SOC_DAPM_AIF_IN("DMIC", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), +SND_SOC_DAPM_AIF_IN("DMIC_ENA", NULL, 0, WM8962_PWR_MGMT_1, 10, 0), SND_SOC_DAPM_ADC("ADCL", "Capture", WM8962_PWR_MGMT_1, 3, 0), SND_SOC_DAPM_ADC("ADCR", "Capture", WM8962_PWR_MGMT_1, 2, 0), @@ -2769,18 +2769,18 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "MICBIAS", NULL, "SYSCLK" }, - { "DMIC", NULL, "DMICDAT" }, + { "DMIC_ENA", NULL, "DMICDAT" }, { "ADCL", NULL, "SYSCLK" }, { "ADCL", NULL, "TOCLK" }, { "ADCL", NULL, "MIXINL" }, - { "ADCL", NULL, "DMIC" }, + { "ADCL", NULL, "DMIC_ENA" }, { "ADCL", NULL, "DSP2" }, { "ADCR", NULL, "SYSCLK" }, { "ADCR", NULL, "TOCLK" }, { "ADCR", NULL, "MIXINR" }, - { "ADCR", NULL, "DMIC" }, + { "ADCR", NULL, "DMIC_ENA" }, { "ADCR", NULL, "DSP2" }, { "STL", "Left", "ADCL" }, -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ASoC: Support a wider range of sample rates on Speyside WM8962
by Mark Brown 23 Sep '11

23 Sep '11

As we've only got one audio interface and it is symmetric we can just set SYSCLK based on the sample rate requested by the application layer. Provide a default so bypass paths work before audio playback. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside_wm8962.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index 3820a6b..98b28ea 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -16,6 +16,8 @@ #include "../codecs/wm8962.h" +static int sample_rate = 44100; + static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) @@ -31,13 +33,13 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, 32768, - 44100 * 512); + sample_rate * 512); if (ret < 0) pr_err("Failed to start FLL: %d\n", ret); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, - 44100 * 512, + sample_rate * 512, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); @@ -109,6 +111,8 @@ static int speyside_wm8962_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + sample_rate = params_rate(params); + return 0; } -- 1.7.6.3

1 0

[alsa-devel] [PATCH] sound/usb/card.c: clear chip->probing on error exit
by Thomas Pfaff 23 Sep '11

23 Sep '11

The Terratec Aureon 5.1 USB sound card support is broken since kernel 2.6.39. 2.6.39 introduced power management support for USB sound cards that added a probing flag in struct snd_usb_audio. During the probe of the card it gives following error message : usb 7-2: new full speed USB device number 2 using uhci_hcd cannot find UAC_HEADER snd-usb-audio: probe of 7-2:1.3 failed with error -5 input: USB Audio as /devices/pci0000:00/0000:00:1d.1/usb7/7-2/7-2:1.3/input/input6 generic-usb 0003:0CCD:0028.0001: input: USB HID v1.00 Device [USB Audio] on usb-0000:00:1d.1-2/input3 I can not comment about that "cannot find UAC_HEADER" error, but until 2.6.38 the card worked anyway. With 2.6.39 chip->probing remains 1 on error exit, and any later ioctl stops in snd_usb_autoresume with -ENODEV. Signed-off-by: Thomas Pfaff <tpfaff(a)gmx.net> --- diff -urp a/sound/usb/card.c b/sound/usb/card.c --- a/sound/usb/card.c 2011-09-22 08:42:29.138247328 +0200 +++ b/sound/usb/card.c 2011-09-22 09:02:09.382079980 +0200 @@ -532,6 +532,7 @@ snd_usb_audio_probe(struct usb_device *d __error: if (chip && !chip->num_interfaces) snd_card_free(chip->card); + chip->probing = 0; mutex_unlock(&register_mutex); __err_val: return NULL;

2 1

[alsa-devel] ASoC updates for 3.2
by Mark Brown 23 Sep '11

23 Sep '11

The following changes since commit f648de832dbf6d1947ce5a7c0ed24a3a71d8545b: Merge branch 'for-3.1' into for-3.2 (2011-09-20 12:59:35 +0100) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Axel Lin (5): ASoC: Staticise bf5xx_pcm_ac97_new() ASoC: Staticise bf5xx_pcm_i2s_new() ASoC: Staticise jz4740_pcm_new() ASoC: Staticise nuc900_dma_getposition() ASoC: Add Kconfig and Makefile entries for rt5631 codec Bas Vermeulen (2): ASoC: 88pm860x-codec - Allow independent use of both I2S playback and capture ASoC: 88pm860x-codec - reset the codec correctly Dong Aisheng (1): ASoC: soc-core: symmetry checking for each DAIs separately Mark Brown (10): ASoC: Add WM5100 driver ASoC: Trace and collect statistics for DAPM graph walking ASoC: Factor out per-widget DAPM power checks ASoC: Ensure we generate a driver name Merge branch 'for-3.1' into for-3.2 ASoC: Include delay.h in 88pm860x Merge branch 'peter/topic/for-mark/mcpdm_for-3.2' of git://gitorious.org/omap-audio/linux-audio into for-3.2 ASoC: Dynamically manage DBVDD2 and DBVDD3 on WM5100 ASoC: Add another DAPM stat for neighbour checks ASoC: Add missed BCLK rate to WM5100 driver Misael Lopez Cruz (1): ASoC: omap-mcpdm: Replace legacy driver Peter Ujfalusi (16): ASoC: tpa6130a2: Model support cleanup ASoC: tpa6130a2: Remove model_id from platform data OMAP: McPDM: Convert McPDM device to omap_device OMAP4: hwmod: enable mcpdm hwmod device. ASoC: omap-mcpdm: Fix threshold and dma configuration ASoC: OMAP4: McPDM: Convert to hwmod/omap_device ASoC: twl6040: Lower the power on gain values at startup ASoC: twl6040: Fix comments for register names ASoC: twl6040: Remove strings "NULL" from DAPM route ASoC: twl6040: Introduce SW only shadow register ASoC: twl6040: Earphone path correction ASoC: twl6040: Use consistent names for Handsfree path ASoC: twl6040: Use consistent names for Headset path ASoC: twl6040: Support for AUX L/R output ASoC: twl6040/sdp4430: Change legacy DAI name ASoC/MFD: twl6040: Combine bit definitions for Headset control registers johnnyhsu(a)realtek.com (1): ASoC: Add driver for rt5631 arch/arm/mach-omap2/board-rx51-peripherals.c | 1 - arch/arm/mach-omap2/devices.c | 33 + arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 +- arch/arm/plat-omap/devices.c | 36 - include/linux/mfd/twl6040.h | 13 +- include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 6 + include/sound/soc.h | 3 +- include/sound/tpa6130a2-plat.h | 6 - include/sound/wm5100.h | 59 + include/trace/events/asoc.h | 25 + sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/codecs/88pm860x-codec.c | 14 +- sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/rt5631.c | 1789 +++++++++ sound/soc/codecs/rt5631.h | 701 ++++ sound/soc/codecs/tpa6130a2.c | 12 +- sound/soc/codecs/twl6040.c | 223 +- sound/soc/codecs/wm5100-tables.c | 1530 ++++++++ sound/soc/codecs/wm5100.c | 2641 +++++++++++++ sound/soc/codecs/wm5100.h | 5146 ++++++++++++++++++++++++++ sound/soc/jz4740/jz4740-pcm.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 470 --- sound/soc/omap/mcpdm.h | 153 - sound/soc/omap/omap-mcpdm.c | 442 ++- sound/soc/omap/omap-mcpdm.h | 95 + sound/soc/omap/sdp4430.c | 6 +- sound/soc/soc-core.c | 18 +- sound/soc/soc-dapm.c | 133 +- sound/soc/soc-pcm.c | 40 +- 34 files changed, 12671 insertions(+), 951 deletions(-) create mode 100644 include/sound/wm5100.h create mode 100644 sound/soc/codecs/rt5631.c create mode 100644 sound/soc/codecs/rt5631.h create mode 100644 sound/soc/codecs/wm5100-tables.c create mode 100644 sound/soc/codecs/wm5100.c create mode 100644 sound/soc/codecs/wm5100.h delete mode 100644 sound/soc/omap/mcpdm.c delete mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.h

2 1

[alsa-devel] [PATCH 1/1] HDA - Add Independent Headphone for all models of ad1988/ad1989
by Raymond Yau 23 Sep '11

23 Sep '11

- Add "AD198x Headphone" playback device for independent headphone playback while playing 7.1 surround using rear panel audio jacks. - Remove "6stack-dig-fp" model since "Headphone Playback Volume" control using DAC0 instead of DAC1 (HDA_FRONT) was already added to all models. - Add "Independent HP" switch to enable/disable this playback device. When the switch is OFF, headphone use "copy front" mode to get the front channel as the green jack. When the switch is ON, you can play stereo sound through "AD198x Headphone" device to headphone while playing 7.1 surround sound through "AD198x Analog" device. The switch cannot be changed when either "AD198x Headphone" or "AD198X Analog" is open.

2 1

Re: [alsa-devel] [PATCH] alsa: 6fire: don't use custom hex_to_bin()
by Takashi Iwai 23 Sep '11

23 Sep '11

At Fri, 23 Sep 2011 14:32:11 +0300, Andy Shevchenko wrote: > > Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: alsa-devel(a)alsa-project.org Applied now. Thanks. Takashi > --- > sound/usb/6fire/firmware.c | 25 ++++++++++++------------- > 1 files changed, 12 insertions(+), 13 deletions(-) > > diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c > index 3ebbdec..3b5f517 100644 > --- a/sound/usb/6fire/firmware.c > +++ b/sound/usb/6fire/firmware.c > @@ -17,6 +17,7 @@ > #include <linux/firmware.h> > #include <linux/module.h> > #include <linux/bitrev.h> > +#include <linux/kernel.h> > > #include "firmware.h" > #include "chip.h" > @@ -60,21 +61,19 @@ struct ihex_record { > unsigned int txt_offset; /* current position in txt_data */ > }; > > -static u8 usb6fire_fw_ihex_nibble(const u8 n) > -{ > - if (n >= '0' && n <= '9') > - return n - '0'; > - else if (n >= 'A' && n <= 'F') > - return n - ('A' - 10); > - else if (n >= 'a' && n <= 'f') > - return n - ('a' - 10); > - return 0; > -} > - > static u8 usb6fire_fw_ihex_hex(const u8 *data, u8 *crc) > { > - u8 val = (usb6fire_fw_ihex_nibble(data[0]) << 4) | > - usb6fire_fw_ihex_nibble(data[1]); > + u8 val = 0; > + int hval; > + > + hval = hex_to_bin(data[0]); > + if (hval >= 0) > + val |= (hval << 4); > + > + hval = hex_to_bin(data[1]); > + if (hval >= 0) > + val |= hval; > + > *crc += val; > return val; > } > -- > 1.7.6.3 >

1 0

[alsa-devel] [PATCH] ASoC: twl6040: No need to read the INTID register
by Peter Ujfalusi 23 Sep '11

23 Sep '11

Since our irq handler has been called, it is granted, that the reason was either PLUGINT, or UNPLUGINT. The INTID register has been checked in the MFD part of twl6040 driver (twl6040-irq.c). We have no reason to read from chip again here. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 9 ++------- 1 files changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index f924827..d5c4bb4 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -747,15 +747,10 @@ static void twl6040_accessory_work(struct work_struct *work) static irqreturn_t twl6040_audio_handler(int irq, void *data) { struct snd_soc_codec *codec = data; - struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - u8 intid; - - intid = twl6040_reg_read(twl6040, TWL6040_REG_INTID); - if ((intid & TWL6040_PLUGINT) || (intid & TWL6040_UNPLUGINT)) - queue_delayed_work(priv->workqueue, &priv->delayed_work, - msecs_to_jiffies(200)); + queue_delayed_work(priv->workqueue, &priv->delayed_work, + msecs_to_jiffies(200)); return IRQ_HANDLED; } -- 1.7.6.1

2 1

Re: [alsa-devel] [PATCH 1/2] ASoC: Staticize rt5631_dai
by Mark Brown 23 Sep '11

23 Sep '11

On Fri, Sep 23, 2011 at 04:22:07PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied both, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: Add missed free_irq in wm5100_remove and wm5100_probe error path
by Mark Brown 23 Sep '11

23 Sep '11

On Fri, Sep 23, 2011 at 01:23:10PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: Remove unneeded mutex_init in wl1273_probe()
by Mark Brown 23 Sep '11

23 Sep '11

On Fri, Sep 23, 2011 at 01:10:57PM +0800, Axel Lin wrote: > Since f0fba2ad "ASoC: multi-component - ASoC Multi-Component Support", > snd_soc_register_codec() now does all the codec list and mutex init. > Thus don't need to call mutex_init(&codec->mutex) in wl1273_probe() any more. Applied, thanks.

1 0

[alsa-devel] [PATCH, resend] ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
by Jarkko Nikula 23 Sep '11

23 Sep '11

Attempt to change McBSP CLKS source while another stream is active is not safe after commit d135865 ("OMAP: McBSP: implement functional clock switching via clock framework") in 2.6.37. CLKS parent clock switching using clock framework have to idle the McBSP before switching and then activate it again. This short break can cause a DMA transaction error to already running stream which halts and recovers only by closing and restarting the stream. This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a severe timeout error and does BUG_ON(). Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk if the McBSP is already active. This test should have been here just from the beginning anyway. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: stable(a)kernel.org Acked-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- For 2.6.37 and onward. Mark: I didn't add your acked-by even you acked this earlier: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-June/040717.html Just in case if this go through your signed-off during kernel.org hazzle. --- sound/soc/omap/omap-mcbsp.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index ebcc2d4..478d607 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,6 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + if (mcbsp_data->active) + if (freq == mcbsp_data->in_freq) + return 0; + else + return -EBUSY; + /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || -- 1.7.5.4

2 1

Re: [alsa-devel] [PATCH] ASoC: Add driver for rt5631
by Mark Brown 23 Sep '11

23 Sep '11

On Thu, Sep 22, 2011 at 09:48:55AM +0800, johnnyhsu wrote: > Besides, could you show me the hint or address where I can post questions > about driver? I wonder how to work between modem and codec and listen to > others' experience. The alsa-devel list is the place for discussions like that.

1 0

[alsa-devel] [PATCHv2] ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
by Jarkko Nikula 23 Sep '11

23 Sep '11

Attempt to change McBSP CLKS source while another stream is active is not safe after commit d135865 ("OMAP: McBSP: implement functional clock switching via clock framework") in 2.6.37. CLKS parent clock switching using clock framework have to idle the McBSP before switching and then activate it again. This short break can cause a DMA transaction error to already running stream which halts and recovers only by closing and restarting the stream. This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a severe timeout error and does BUG_ON(). Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk if the McBSP is already active. This test should have been here just from the beginning anyway. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: stable(a)kernel.org --- v2: - Returns -EBUSY from omap_mcbsp_dai_set_dai_sysclk if the input clock rate doesn't match with another stream's configuration and 0 otherwise. For 2.6.37 and onward. This issue is valid to those machine drivers that use the McBSP as a DAI link master and are configuring the CLKS source (like sound/soc/omap/omap3pandora.c in mainline). --- sound/soc/omap/omap-mcbsp.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 07b7723..4b82290 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,6 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + if (mcbsp_data->active) + if (freq == mcbsp_data->in_freq) + return 0; + else + return -EBUSY; + /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || -- 1.7.4.4

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[alsa-devel] snd_pcm_hw_rule_noresample
by Raymond Yau 23 Sep '11

23 Sep '11

60fdc82f96bff2e3a31e9fcf8a9503f5a37c1f85 ALSA: pcm: add snd_pcm_hw_rule_noresample() Add a helper function to allow drivers to disable hardware resampling when the application has specified the SNDRV_PCM_HW_PARAMS_NORESAMPLE flag. Refer to commit and the three corresponding patch with ymfpci, emu10k1, via82xx, it seem to be a regression according to http://www.alsa-project.org/alsa-doc/alsa-lib/group___p_c_m___h_w___params.… int snd_pcm_hw_params_set_rate_resample ( snd_pcm_t * pcm, snd_pcm_hw_params_t * params, unsigned int val ) Restrict a configuration space to contain only real hardware rates. ymfpci and emu10k1 have SNDRV_PCM_RATE_CONTINUOUS and support 8000Hz to 48000Hz after this patch and if application using snd_pcm_hw_params_set_rate_resample(,pcm , params ,0) why the supported rate is only 48000Hz ? but snd_pcm_hw_params_get_rate_min() is still 8000Hz This patch won't affect the accuracy of the playback position of the ymfpci attach a test program which print the the playback position and the result ./alsa_test hw:0 Hardware PCM card 0 'Yamaha DS-1 (YMF724F)' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 16383 period_size : 5461 period_time : 123832 tstamp_mode : ENABLE period_step : 1 avail_min : 5461 period_event : 0 start_threshold : 1 stop_threshold : 16383 silence_threshold: 0 silence_size : 0 boundary : 2147352576 appl_ptr : 0 hw_ptr : 0 pcm write 5461 pcm write 5461 pcm write 5461 0 0 1 0 2 0 3 0 4 0 5 0 6 0 7 0 8 0 9 0 10 0 11 235 12 235 13 235 14 235 15 235 16 470 17 470 18 470 19 470 20 470 21 705 22 705 23 705 24 705 25 705 26 940 27 940 28 940 29 940 30 940 31 940 32 1175 33 1175 34 1175 35 1175 36 1175 37 1411 38 1411 39 1411 40 1411 41 1411 42 1646 43 1646 44 1646 45 1646 46 1646 47 1646 48 1881 49 1881 50 1881 51 1881 52 1881 53 2116 54 2116 55 2116 56 2116 57 2116 58 2351 59 2351 60 2351 61 2351 62 2351 63 2587 64 2587 65 2587 66 2587 67 2587 68 2587 69 2822 70 2822 71 2822 72 2822 73 2822 74 3057 75 3057 76 3057 77 3057 78 3057 79 3292 80 3292 81 3292 82 3292 83 3292 84 3527 85 3527 86 3527 87 3527 88 3527 89 3527 90 3763 91 3763 92 3763 93 3763 94 3763 95 3998 96 3998 97 3998 98 3998 99 3998 100 4233 101 4233 102 4233 103 4233 104 4233 105 4233 106 4468 107 4468 108 4468 109 4468 110 4468 111 4703 112 4703 113 4703 114 4703 115 4703 116 4939 117 4939 118 4939 119 4939 120 4939 121 5174 122 5174 123 5174 124 5174 125 5174 126 5174 127 5409 128 5409 129 5409 130 5409 131 5409 132 5644 133 5644 134 5644 135 5644 136 5644 137 5879 138 5879 139 5879 140 5879 141 5879 142 6115 143 6115 144 6115 145 6115 146 6115 147 6115 148 6350 149 6350 150 6350 151 6350 152 6350 153 6585 154 6585 155 6585 156 6585 157 6585 158 6820 159 6820 160 6820 161 6820 162 6820 163 6820 164 7055 165 7055 166 7055 167 7055 168 7055 169 7291 170 7291 171 7291 172 7291 173 7291 174 7526 175 7526 176 7526 177 7526 178 7526 179 7761 180 7761 181 7761 182 7761 183 7761 184 7761 185 7996 186 7996 187 7996 188 7996 189 7996 190 8231 191 8231 192 8231 193 8231 194 8231 195 8467 196 8467 197 8467 198 8467 199 8467 200 8702 201 8702 202 8702 203 8702 204 8702 205 8702 206 8937 207 8937 208 8937 209 8937 210 8937 211 9172 212 9172 213 9172 214 9172 215 9172 216 9407 217 9407 218 9407 219 9407 220 9407 221 9407 222 9643 223 9643 224 9643 225 9643 226 9643 227 9878 228 9878 229 9878 230 9878 231 9878 232 10113 233 10113 234 10113 235 10113 236 10113 237 10348 238 10348 239 10348 240 10348 241 10348 242 10348 243 10583 244 10583 245 10583 246 10583 247 10583 248 10819 249 10819 250 10819 251 10819 252 10819 253 11054 254 11054 255 11054 256 11054 257 11054 258 11289 259 11289 260 11289 261 11289 262 11289 263 11289 264 11524 265 11524 266 11524 267 11524 268 11524 269 11759 270 11759 271 11759 272 11759 273 11759 274 11995 275 11995 276 11995 277 11995 278 11995 279 11995 280 12230 281 12230 282 12230 283 12230 284 12230 285 12465 286 12465 287 12465 288 12465 289 12465 290 12700 291 12700 292 12700 293 12700 294 12700 295 12935 296 12935 297 12935 298 12935 299 12935 300 12935 301 13171 302 13171 303 13171 304 13171 305 13171 306 13406 307 13406 308 13406 309 13406 310 13406 311 13641 312 13641 313 13641 314 13641 315 13641 316 13876 317 13876 318 13876 319 13876 320 13876 321 13876 322 14111 323 14111 324 14111 325 14111 326 14111 327 14347 328 14347 329 14347 330 14347 331 14347 332 14582 333 14582 334 14582 335 14582 336 14582 337 14582 338 14817 339 14817 340 14817 341 14817 342 14817 343 15052 344 15052 345 15052 346 15052 347 15052 348 15287 349 15287 350 15287 351 15287 352 15287 353 15523 354 15523 355 15523 356 15523 357 15523 358 15523 359 15758 360 15758 361 15758 362 15758 363 15758 364 15993 365 15993 366 15993 367 15993 368 15993 369 16228 370 16228 371 16228 372 16228 373 16228 374 -32

2 4

Re: [alsa-devel] [patch 2/2] ALSA: hdspm - cleanup __user tags in ioctl()
by Takashi Iwai 23 Sep '11

23 Sep '11

At Fri, 23 Sep 2011 09:25:05 +0300, Dan Carpenter wrote: > > This makes the code cleaner and silences a Sparse complaint: > sound/pci/rme9652/hdspm.c:6341:23: warning: incorrect type in assignment (incompatible argument 4 (different address spaces)) > sound/pci/rme9652/hdspm.c:6341:23: expected int ( *ioctl )( ... ) > sound/pci/rme9652/hdspm.c:6341:23: got int ( static [toplevel] *<noident> )( ... ) > sound/pci/rme9652/hdspm.c:6102:44: warning: dereference of noderef expression > sound/pci/rme9652/hdspm.c:6225:50: warning: dereference of noderef expression > sound/pci/rme9652/hdspm.c:6264:50: warning: dereference of noderef expression > sound/pci/rme9652/hdspm.c:6283:50: warning: dereference of noderef expression > sound/pci/rme9652/hdspm.c:6289:59: warning: dereference of noderef expression > > Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Applied now. Thanks. Takashi > diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c > index bf438d1..6e2f7ef 100644 > --- a/sound/pci/rme9652/hdspm.c > +++ b/sound/pci/rme9652/hdspm.c > @@ -6097,7 +6097,7 @@ static inline int copy_u32_le(void __user *dest, void __iomem *src) > } > > static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > - unsigned int cmd, unsigned long __user arg) > + unsigned int cmd, unsigned long arg) > { > void __user *argp = (void __user *)arg; > struct hdspm *hdspm = hw->private_data; > @@ -6222,7 +6222,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > info.line_out = hdspm_line_out(hdspm); > info.passthru = 0; > spin_unlock_irq(&hdspm->lock); > - if (copy_to_user((void __user *) arg, &info, sizeof(info))) > + if (copy_to_user(argp, &info, sizeof(info))) > return -EFAULT; > break; > > @@ -6261,7 +6261,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > break; > } > > - if (copy_to_user((void __user *) arg, &status, sizeof(status))) > + if (copy_to_user(argp, &status, sizeof(status))) > return -EFAULT; > > > @@ -6280,13 +6280,13 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > if (hdspm->tco) > hdspm_version.addons |= HDSPM_ADDON_TCO; > > - if (copy_to_user((void __user *) arg, &hdspm_version, > + if (copy_to_user(argp, &hdspm_version, > sizeof(hdspm_version))) > return -EFAULT; > break; > > case SNDRV_HDSPM_IOCTL_GET_MIXER: > - if (copy_from_user(&mixer, (void __user *)arg, sizeof(mixer))) > + if (copy_from_user(&mixer, argp, sizeof(mixer))) > return -EFAULT; > if (copy_to_user((void __user *)mixer.mixer, hdspm->mixer, > sizeof(struct hdspm_mixer))) >

1 0

Re: [alsa-devel] [patch 1/2] ALSA: hdspm - potential info leak in snd_hdspm_hwdep_ioctl()
by Takashi Iwai 23 Sep '11

23 Sep '11

At Fri, 23 Sep 2011 09:24:21 +0300, Dan Carpenter wrote: > > Smatch has a new check for Rosenberg type information leaks where > structs are copied to the user with uninitialized stack data in them. > > The status struct has a hole in it, and on some paths not all the > members were initialized. > > struct hdspm_status { > unsigned char card_type; /* 0 1 */ > /* XXX 3 bytes hole, try to pack */ > enum hdspm_syncsource autosync_source; /* 4 4 */ > long long unsigned int card_clock; /* 8 8 */ > > The hdspm_version struct had holes in it as well. > > struct hdspm_version { > unsigned char card_type; /* 0 1 */ > char cardname[20]; /* 1 20 */ > /* XXX 3 bytes hole, try to pack */ > unsigned int serial; /* 24 4 */ > short unsigned int firmware_rev; /* 28 2 */ > /* XXX 2 bytes hole, try to pack */ > int addons; /* 32 4 */ > > Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> Applied now. Thanks. Takashi > diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c > index 214110d..bf438d1 100644 > --- a/sound/pci/rme9652/hdspm.c > +++ b/sound/pci/rme9652/hdspm.c > @@ -6227,6 +6227,8 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > break; > > case SNDRV_HDSPM_IOCTL_GET_STATUS: > + memset(&status, 0, sizeof(status)); > + > status.card_type = hdspm->io_type; > > status.autosync_source = hdspm_autosync_ref(hdspm); > @@ -6266,6 +6268,8 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, > break; > > case SNDRV_HDSPM_IOCTL_GET_VERSION: > + memset(&hdspm_version, 0, sizeof(hdspm_version)); > + > hdspm_version.card_type = hdspm->io_type; > strncpy(hdspm_version.cardname, hdspm->card_name, > sizeof(hdspm_version.cardname)); >

1 0

[alsa-devel] ASoC trees while master is down
by Mark Brown 23 Sep '11

23 Sep '11

Due to the continued outage of kernel.org I'm currently publishing my for-whatever branches on: git://opensource.wolfsonmicro.com/linux-2.6-asoc If you need current code please look there.

2 2

[alsa-devel] [PATCH] ASoC: Add missed BCLK rate to WM5100 driver
by Mark Brown 22 Sep '11

22 Sep '11

Reported-by: Axel Lin <axel.lin(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 443d76d..cb940a8 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1511,6 +1511,7 @@ static int wm5100_bclk_rates_dat[WM5100_NUM_BCLK_RATES] = { 96000, 128000, 192000, + 256000, 384000, 512000, 768000, -- 1.7.6.3

1 0

[alsa-devel] [PATCH v2 00/10] ASoC: twl6040: Cleanups and fixes
by Peter Ujfalusi 22 Sep '11

22 Sep '11

Hello, Changes since v1 [1]: - Left out patches already applied by Mark - Description on patch 01 changed to describe more precisely the need to lower the chip default gains after power on. - Extended the commit message for patch 02 to explain the change. - Added Acked-by from Liam to the patches. Short summary for v2 series: Code cleanup, and fixes for the twl6040 codec driver. Main changes: - codec initialization changes - Correction for the Earphone path - AUX L/R output support - Headset/Handsfree control name changes - Headset DC offset workaround The series has been created on top of: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-3.2 branch + OMAP McPDM changes [2] [1] first series: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/043847.… [2] OMAP McPDM changes: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-September/044172.… Regards, Peter --- Peter Ujfalusi (10): ASoC: twl6040: Lower the power on gain values at startup ASoC: twl6040: Fix comments for register names ASoC: twl6040: Remove strings "NULL" from DAPM route ASoC: twl6040: Introduce SW only shadow register ASoC: twl6040: Earphone path correction ASoC: twl6040: Use consistent names for Handsfree path ASoC: twl6040: Use consistent names for Headset path ASoC: twl6040: Support for AUX L/R output ASoC: twl6040/sdp4430: Change legacy DAI name ASoC/MFD: twl6040: Combine bit definitions for Headset control registers include/linux/mfd/twl6040.h | 13 +-- sound/soc/codecs/twl6040.c | 223 ++++++++++++++++++++++++++----------------- sound/soc/omap/sdp4430.c | 2 +- 3 files changed, 140 insertions(+), 98 deletions(-) -- 1.7.6.1

2 11

Re: [alsa-devel] Using the timer
by Clemens Ladisch 22 Sep '11

22 Sep '11

Ilya Dmitrichenko wrote: > So you are saying that PCM slave timer can be used in some > way, but it would be clocked at the sampling rate, right? We get an interrupt at every period boundary, so the timer would run at, e.g., one 128th of the sampling rate. > Hm, so 'snd-seq-timer' is essentially hooked to the host cpu > clock (be that TSC/RTC/HPET) ? In the default setting, yes. Regards, Clemens

1 0

[alsa-devel] [PATCH 1/2] fm801: Fix double free in case of error in tuner detection
by Ben Hutchings 22 Sep '11

22 Sep '11

Commit 9676001559fce06e37c7dc230ab275f605556176 ("ALSA: fm801: add error handling if auto-detect fails") added incorrect error handling. Once we have successfully called snd_device_new(), the cleanup function fm801_free() will automatically be called by snd_card_free() and we must *not* also call fm801_free() directly. Reported-by: Hor Jiun Shyong <jiunshyong(a)gmail.com> References: http://bugs.debian.org/641946 Signed-off-by: Ben Hutchings <ben(a)decadent.org.uk> Cc: stable(a)kernel.org [v3.0+] --- sound/pci/fm801.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index a7ec703..0dd8963 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1236,7 +1236,6 @@ static int __devinit snd_fm801_create(struct snd_card *card, (tea575x_tuner & TUNER_TYPE_MASK) < 4) { if (snd_tea575x_init(&chip->tea)) { snd_printk(KERN_ERR "TEA575x radio not found\n"); - snd_fm801_free(chip); return -ENODEV; } } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) { @@ -1251,7 +1250,6 @@ static int __devinit snd_fm801_create(struct snd_card *card, } if (tea575x_tuner == 4) { snd_printk(KERN_ERR "TEA575x radio not found\n"); - snd_fm801_free(chip); return -ENODEV; } } -- 1.7.5.4

2 1

[alsa-devel] [PATCH 2/2] fm801: Gracefully handle failure of tuner auto-detect
by Ben Hutchings 22 Sep '11

22 Sep '11

Commit 9676001559fce06e37c7dc230ab275f605556176 ("ALSA: fm801: add error handling if auto-detect fails") seems to break systems that were previously working without a tuner. As a bonus, this should fix init and cleanup for the case where the tuner is explicitly disabled. Reported-and-tested-by: Hor Jiun Shyong <jiunshyong(a)gmail.com> References: http://bugs.debian.org/641946 Signed-off-by: Ben Hutchings <ben(a)decadent.org.uk> Cc: stable(a)kernel.org [v3.0+] --- sound/pci/fm801.c | 13 ++++++++++--- 1 files changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 0dd8963..ecce948 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -68,6 +68,7 @@ MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only)."); +#define TUNER_DISABLED (1<<3) #define TUNER_ONLY (1<<4) #define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) @@ -1150,7 +1151,8 @@ static int snd_fm801_free(struct fm801 *chip) __end_hw: #ifdef CONFIG_SND_FM801_TEA575X_BOOL - snd_tea575x_exit(&chip->tea); + if (!(chip->tea575x_tuner & TUNER_DISABLED)) + snd_tea575x_exit(&chip->tea); #endif if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1250,10 +1252,15 @@ static int __devinit snd_fm801_create(struct snd_card *card, } if (tea575x_tuner == 4) { snd_printk(KERN_ERR "TEA575x radio not found\n"); - return -ENODEV; + chip->tea575x_tuner = TUNER_DISABLED; } } - strlcpy(chip->tea.card, snd_fm801_tea575x_gpios[(tea575x_tuner & TUNER_TYPE_MASK) - 1].name, sizeof(chip->tea.card)); + if (!(chip->tea575x_tuner & TUNER_DISABLED)) { + strlcpy(chip->tea.card, + snd_fm801_tea575x_gpios[(tea575x_tuner & + TUNER_TYPE_MASK) - 1].name, + sizeof(chip->tea.card)); + } #endif *rchip = chip; -- 1.7.5.4

1 0

[alsa-devel] [PATCH 2/4] src/pcm/pcm_mmap.c: add missing free
by Julia Lawall 22 Sep '11

22 Sep '11

From: Julia Lawall <julia(a)diku.dk> Something that is allocated using calloc is not freed on an error path. Signed-off-by: Julia Lawall <julia(a)diku.dk> Signed-off-by: Suman Saha <sumsaha(a)gmail.com> --- Compile-tested only. src/pcm/pcm_mmap.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/src/pcm/pcm_mmap.c b/src/pcm/pcm_mmap.c index 4621fe6..2a009c8 100644 --- a/src/pcm/pcm_mmap.c +++ b/src/pcm/pcm_mmap.c @@ -320,8 +320,11 @@ int snd_pcm_mmap(snd_pcm_t *pcm) snd_pcm_channel_info_t *i = &pcm->mmap_channels[c]; i->channel = c; err = snd_pcm_channel_info(pcm, i); - if (err < 0) + if (err < 0) { + free(pcm->mmap_channels); + pcm->mmap_channels = NULL; return err; + } } for (c = 0; c < pcm->channels; ++c) { snd_pcm_channel_info_t *i = &pcm->mmap_channels[c];

2 3

Re: [alsa-devel] Using the timer
by Clemens Ladisch 22 Sep '11

22 Sep '11

Ilya Dmitrichenko wrote: > On 21 September 2011 13:54, Clemens Ladisch <clemens(a)ladisch.de> wrote: > > What hardware timer? > > I meant the clock of the soundcard. The HDSP does have a sample clock, but it cannot be used independently from a running PCM stream. Using this PCM slave timer in the sequencer is possible, but the period length is likely to be much larger than the desired sequencer timestamp granularity. What exactly do you want to do? Regards, Clemens

1 0

[alsa-devel] [PATCH 1/4] src/pcm/pcm_rate.c: add missing free
by Julia Lawall 22 Sep '11

22 Sep '11

From: Julia Lawall <julia(a)diku.dk> Something that is allocated using calloc is not freed on one or more error paths. Signed-off-by: Julia Lawall <julia(a)diku.dk> Signed-off-by: Suman Saha <sumsaha(a)gmail.com> --- Compile-tested only. src/pcm/pcm_rate.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 70e30e5..eb35e4a 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1392,11 +1392,13 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, } else { SNDERR("Invalid type for rate converter"); snd_pcm_close(pcm); + free(rate); return -EINVAL; } if (err < 0) { SNDERR("Cannot find rate converter"); snd_pcm_close(pcm); + free(rate); return -ENOENT; } #else @@ -1405,6 +1407,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, err = open_func(SND_PCM_RATE_PLUGIN_VERSION, &rate->obj, &rate->ops); if (err < 0) { snd_pcm_close(pcm); + free(rate); return err; } #endif @@ -1413,6 +1416,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, ! rate->ops.input_frames || ! rate->ops.output_frames) { SNDERR("Inproper rate plugin %s initialization", type); snd_pcm_close(pcm); + free(rate); return err; }

2 2

[alsa-devel] [GIT PULL] OMAP McPDM updates for 3.2
by Péter Ujfalusi 22 Sep '11

22 Sep '11

Hello Mark, as we discussed you can pull the OMAP McPDM updates. As for reference: ASoC: OMAP4: McPDM: Fix legacy support [4 patch] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/042401.html ASoC: omap-mcpdm: Replace legacy driver: http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/043317.html All received acks Regards, Péter The following changes since commit 0722d055ac2236da4e319d22a99c9f7e82dbdd5d: ASoC: tpa6130a2: Remove model_id from platform data (2011-09-21 16:07:45 +0100) are available in the git repository at: git://gitorious.org/omap-audio/linux-audio.git peter/topic/for-mark/mcpdm_for-3.2 Misael Lopez Cruz (1): ASoC: omap-mcpdm: Replace legacy driver Peter Ujfalusi (4): OMAP: McPDM: Convert McPDM device to omap_device OMAP4: hwmod: enable mcpdm hwmod device. ASoC: omap-mcpdm: Fix threshold and dma configuration ASoC: OMAP4: McPDM: Convert to hwmod/omap_device arch/arm/mach-omap2/devices.c | 33 ++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 +- arch/arm/plat-omap/devices.c | 36 --- sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 470 ---------------------------- sound/soc/omap/mcpdm.h | 153 --------- sound/soc/omap/omap-mcpdm.c | 442 ++++++++++++++++++++------ sound/soc/omap/omap-mcpdm.h | 95 ++++++ sound/soc/omap/sdp4430.c | 4 +- 9 files changed, 472 insertions(+), 765 deletions(-) delete mode 100644 sound/soc/omap/mcpdm.c delete mode 100644 sound/soc/omap/mcpdm.h create mode 100644 sound/soc/omap/omap-mcpdm.h

2 1

[alsa-devel] [PATCH] HDA - add "Independent HP" switch for ad1988
by Raymond Yau 22 Sep '11

22 Sep '11

Add "Independent HP" switch for ad1988 - add playback device 2 "AD1988 HP" for 7.1+2 multi streaming playback - add "Independent HP" switch to enable/disable the feature switch is inactive and write protect when device 0 or device 2 is opened - remove 6stack-dig-fp model

2 3

[alsa-devel] [PATCH] ASoC: Include delay.h in 88pm860x
by Mark Brown 22 Sep '11

22 Sep '11

Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/88pm860x-codec.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index df7b4a0..5ca122e 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -15,6 +15,7 @@ #include <linux/platform_device.h> #include <linux/mfd/88pm860x.h> #include <linux/slab.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> -- 1.7.6.3

1 0

[alsa-devel] [PATCH 1/2] ASoC: Add platform data for WM1250 EV1 GPIOs
by Mark Brown 22 Sep '11

22 Sep '11

The WM1250 EV1 has some GPIOs which can be used to control the behaviour at runtime. Request them all if supplied and add a set_bias_level() function to start and stop the clocks. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/wm1250-ev1.h | 27 +++++++++ sound/soc/codecs/wm1250-ev1.c | 121 ++++++++++++++++++++++++++++++++++++++++- 2 files changed, 145 insertions(+), 3 deletions(-) create mode 100644 include/sound/wm1250-ev1.h diff --git a/include/sound/wm1250-ev1.h b/include/sound/wm1250-ev1.h new file mode 100644 index 0000000..7dff828 --- /dev/null +++ b/include/sound/wm1250-ev1.h @@ -0,0 +1,27 @@ +/* + * linux/sound/wm1250-ev1.h - Platform data for WM1250-EV1 + * + * Copyright 2011 Wolfson Microelectronics. PLC. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_WM1250_EV1_H +#define __LINUX_SND_WM1250_EV1_H + +#define WM1250_EV1_NUM_GPIOS 5 + +#define WM1250_EV1_GPIO_CLK_ENA 0 +#define WM1250_EV1_GPIO_CLK_SEL0 1 +#define WM1250_EV1_GPIO_CLK_SEL1 2 +#define WM1250_EV1_GPIO_OSR 3 +#define WM1250_EV1_GPIO_MASTER 4 + + +struct wm1250_ev1_pdata { + int gpios[WM1250_EV1_NUM_GPIOS]; +}; + +#endif diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index 4523c4c..0554b88 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -12,10 +12,59 @@ #include <linux/init.h> #include <linux/module.h> +#include <linux/slab.h> #include <linux/i2c.h> +#include <linux/gpio.h> #include <sound/soc.h> #include <sound/soc-dapm.h> +#include <sound/wm1250-ev1.h> + +static const char *wm1250_gpio_names[WM1250_EV1_NUM_GPIOS] = { + "WM1250 CLK_ENA", + "WM1250 CLK_SEL0", + "WM1250 CLK_SEL1", + "WM1250 OSR", + "WM1250 MASTER", +}; + +struct wm1250_priv { + struct gpio gpios[WM1250_EV1_NUM_GPIOS]; +}; + +static int wm1250_ev1_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm1250_priv *wm1250 = dev_get_drvdata(codec->dev); + int ena; + + if (wm1250) + ena = wm1250->gpios[WM1250_EV1_GPIO_CLK_ENA].gpio; + else + ena = -1; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (ena >= 0) + gpio_set_value_cansleep(ena, 1); + break; + + case SND_SOC_BIAS_OFF: + if (ena >= 0) + gpio_set_value_cansleep(ena, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} static const struct snd_soc_dapm_widget wm1250_ev1_dapm_widgets[] = { SND_SOC_DAPM_ADC("ADC", "wm1250-ev1 Capture", SND_SOC_NOPM, 0, 0), @@ -53,12 +102,66 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { .num_dapm_widgets = ARRAY_SIZE(wm1250_ev1_dapm_widgets), .dapm_routes = wm1250_ev1_dapm_routes, .num_dapm_routes = ARRAY_SIZE(wm1250_ev1_dapm_routes), + + .set_bias_level = wm1250_ev1_set_bias_level, }; +static int __devinit wm1250_ev1_pdata(struct i2c_client *i2c) +{ + struct wm1250_ev1_pdata *pdata = dev_get_platdata(&i2c->dev); + struct wm1250_priv *wm1250; + int i, ret; + + if (!pdata) + return 0; + + wm1250 = kzalloc(sizeof(*wm1250), GFP_KERNEL); + if (!wm1250) { + dev_err(&i2c->dev, "Unable to allocate private data\n"); + ret = -ENOMEM; + goto err; + } + + for (i = 0; i < ARRAY_SIZE(wm1250->gpios); i++) { + wm1250->gpios[i].gpio = pdata->gpios[i]; + wm1250->gpios[i].label = wm1250_gpio_names[i]; + wm1250->gpios[i].flags = GPIOF_OUT_INIT_LOW; + } + wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL0].flags = GPIOF_OUT_INIT_HIGH; + wm1250->gpios[WM1250_EV1_GPIO_CLK_SEL1].flags = GPIOF_OUT_INIT_HIGH; + + ret = gpio_request_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to get GPIOs: %d\n", ret); + goto err_alloc; + } + + dev_set_drvdata(&i2c->dev, wm1250); + + return ret; + +err_alloc: + kfree(wm1250); +err: + return ret; +} + +static void wm1250_ev1_free(struct i2c_client *i2c) +{ + struct wm1250_priv *wm1250 = dev_get_drvdata(&i2c->dev); + + if (wm1250) { + gpio_free_array(wm1250->gpios, ARRAY_SIZE(wm1250->gpios)); + kfree(wm1250); + } +} + static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, const struct i2c_device_id *i2c_id) { - int id, board, rev; + int id, board, rev, ret; + + dev_set_drvdata(&i2c->dev, NULL); board = i2c_smbus_read_byte_data(i2c, 0); if (board < 0) { @@ -76,13 +179,25 @@ static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "revision %d\n", rev + 1); - return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, - &wm1250_ev1_dai, 1); + ret = wm1250_ev1_pdata(i2c); + if (ret != 0) + return ret; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, + &wm1250_ev1_dai, 1); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + wm1250_ev1_free(i2c); + return ret; + } + + return 0; } static int __devexit wm1250_ev1_remove(struct i2c_client *i2c) { snd_soc_unregister_codec(&i2c->dev); + wm1250_ev1_free(i2c); return 0; } -- 1.7.6.3

1 1

[alsa-devel] sound card matrix defaced
by Patrick Shirkey 22 Sep '11

22 Sep '11

Hi, Someone needs to restore the sound card matrix as it has been defaced by spammers. Cheers -- Patrick Shirkey Boost Hardware Ltd

2 2

[alsa-devel] [PATCH] ASoC: Don't force bias on ground referenced devices
by Mark Brown 22 Sep '11

22 Sep '11

Currently we force all devices in the system to be at the same bias level. This is due to concerns about power or pop/click impacts from either ramping VMID or mismatching VMID on the analogue I/O lines between connected devices but does mean we power devices up more often than we really need to. If a device flags idle_bias_off this will usually mean that it's either all digital or ground referenced (in which case the idle and powered bias levels are identical) so this concern does not apply and we can save some power by leaving it off when not needed itself. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6cac045..2bde6b0c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1319,13 +1319,16 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } - /* Force all contexts in the card to the same bias state */ + /* Force all contexts in the card to the same bias state if + * they're not ground referenced. + */ bias = SND_SOC_BIAS_OFF; list_for_each_entry(d, &card->dapm_list, list) if (d->target_bias_level > bias) bias = d->target_bias_level; list_for_each_entry(d, &card->dapm_list, list) - d->target_bias_level = bias; + if (!d->idle_bias_off) + d->target_bias_level = bias; trace_snd_soc_dapm_walk_done(card); -- 1.7.6.3

1 0

[alsa-devel] [PATCH] alsa-plugins: Pulse: only underrun if no more data has been written
by David Henningsson 22 Sep '11

22 Sep '11

With the new/upcoming version of PulseAudio (0.99.x) there is a protocol addition that makes it possible to handle underruns better in the pulse plugin. The attached patch implements that, but it has two flaws I wouldn't mind getting some help with if possible: 1) Since this uses the new API function pa_stream_get_underflow_index, it won't compile with current stable PulseAudio versions, only the upcoming version. 2) So now there are three possibilities for handle_underrun ( 0 = never, 1 = always, 2 = the new improved one that IMO should be used). Since handle_underrun is a bool in the config, it can not be set to "2", only 0 or 1. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 13

[alsa-devel] [PATCH] ASoC: Add another DAPM stat for neighbour checks
by Mark Brown 21 Sep '11

21 Sep '11

The number of times we look at a potentially connected neighbour is just as important as the number of times we actually recurse into looking at that neighbour so also collect that statistic. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + include/trace/events/asoc.h | 7 +++++-- sound/soc/soc-dapm.c | 4 ++++ 3 files changed, 10 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 0e2d017..bb59532 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -540,6 +540,7 @@ struct snd_soc_dapm_widget_list { struct snd_soc_dapm_stats { int power_checks; int path_checks; + int neighbour_checks; }; #endif diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 2e1adf6..ab26f8a 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -226,16 +226,19 @@ TRACE_EVENT(snd_soc_dapm_walk_done, __string( name, card->name ) __field( int, power_checks ) __field( int, path_checks ) + __field( int, neighbour_checks ) ), TP_fast_assign( __assign_str(name, card->name); __entry->power_checks = card->dapm_stats.power_checks; __entry->path_checks = card->dapm_stats.path_checks; + __entry->neighbour_checks = card->dapm_stats.neighbour_checks; ), - TP_printk("%s: %d power checks, %d path checks", __get_str(name), - (int)__entry->power_checks, (int)__entry->path_checks) + TP_printk("%s: checks %d power, %d path, %d neighbour", + __get_str(name), (int)__entry->power_checks, + (int)__entry->path_checks, (int)__entry->neighbour_checks) ); TRACE_EVENT(snd_soc_jack_irq, diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 84d1d79..6cac045 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -677,6 +677,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { + DAPM_UPDATE_STAT(widget, neighbour_checks); + if (path->weak) continue; @@ -732,6 +734,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { + DAPM_UPDATE_STAT(widget, neighbour_checks); + if (path->weak) continue; -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ASoC: Dynamically manage DBVDD2 and DBVDD3 on WM5100
by Mark Brown 21 Sep '11

21 Sep '11

Allow the DBVDD2 and DBVDD3 rails to be powered down when idle, helping fully power down connected devices when idle. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5100.c | 82 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 81 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index a30b0bf..eff252b 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -55,6 +55,8 @@ struct wm5100_priv { struct regulator_bulk_data core_supplies[WM5100_NUM_CORE_SUPPLIES]; struct regulator *cpvdd; + struct regulator *dbvdd2; + struct regulator *dbvdd3; int rev; @@ -804,6 +806,52 @@ static int wm5100_cp_ev(struct snd_soc_dapm_widget *w, } } +static int wm5100_dbvdd_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct regulator *regulator; + int ret; + + switch (w->shift) { + case 2: + regulator = wm5100->dbvdd2; + break; + case 3: + regulator = wm5100->dbvdd3; + break; + default: + BUG(); + return 0; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(regulator); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", + w->shift, ret); + return ret; + } + return ret; + + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(regulator); + if (ret != 0) { + dev_err(codec->dev, "Failed to enable DBVDD%d: %d\n", + w->shift, ret); + return ret; + } + return ret; + + default: + BUG(); + return 0; + } +} + static void wm5100_log_status3(struct snd_soc_codec *codec, int val) { if (val & WM5100_SPK_SHUTDOWN_WARN_EINT) @@ -881,6 +929,10 @@ SND_SOC_DAPM_SUPPLY("CP2", WM5100_MIC_CHARGE_PUMP_1, WM5100_CP2_ENA_SHIFT, 0, SND_SOC_DAPM_SUPPLY("CP2 Active", WM5100_MIC_CHARGE_PUMP_1, WM5100_CP2_BYPASS_SHIFT, 1, wm5100_cp_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DBVDD2", SND_SOC_NOPM, 2, 0, wm5100_dbvdd_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("DBVDD3", SND_SOC_NOPM, 3, 0, wm5100_dbvdd_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("MICBIAS1", WM5100_MIC_BIAS_CTRL_1, WM5100_MICB1_ENA_SHIFT, 0, NULL, 0), @@ -1123,10 +1175,14 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "AIF1RX8", NULL, "SYSCLK" }, { "AIF2RX1", NULL, "SYSCLK" }, + { "AIF2RX1", NULL, "DBVDD2" }, { "AIF2RX2", NULL, "SYSCLK" }, + { "AIF2RX2", NULL, "DBVDD2" }, { "AIF3RX1", NULL, "SYSCLK" }, + { "AIF3RX1", NULL, "DBVDD3" }, { "AIF3RX2", NULL, "SYSCLK" }, + { "AIF3RX2", NULL, "DBVDD3" }, { "AIF1TX1", NULL, "SYSCLK" }, { "AIF1TX2", NULL, "SYSCLK" }, @@ -1138,10 +1194,14 @@ static const struct snd_soc_dapm_route wm5100_dapm_routes[] = { { "AIF1TX8", NULL, "SYSCLK" }, { "AIF2TX1", NULL, "SYSCLK" }, + { "AIF2TX1", NULL, "DBVDD2" }, { "AIF2TX2", NULL, "SYSCLK" }, + { "AIF2TX2", NULL, "DBVDD2" }, { "AIF3TX1", NULL, "SYSCLK" }, + { "AIF3TX1", NULL, "DBVDD3" }, { "AIF3TX2", NULL, "SYSCLK" }, + { "AIF3TX2", NULL, "DBVDD3" }, { "MICBIAS1", NULL, "CP2" }, { "MICBIAS2", NULL, "CP2" }, @@ -2251,12 +2311,26 @@ static int wm5100_probe(struct snd_soc_codec *codec) goto err_core; } + wm5100->dbvdd2 = regulator_get(&i2c->dev, "DBVDD2"); + if (IS_ERR(wm5100->dbvdd2)) { + ret = PTR_ERR(wm5100->dbvdd2); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_cpvdd; + } + + wm5100->dbvdd3 = regulator_get(&i2c->dev, "DBVDD3"); + if (IS_ERR(wm5100->dbvdd3)) { + ret = PTR_ERR(wm5100->dbvdd3); + dev_err(&i2c->dev, "Failed to get DBVDD2: %d\n", ret); + goto err_dbvdd2; + } + ret = regulator_bulk_enable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); if (ret != 0) { dev_err(codec->dev, "Failed to enable core supplies: %d\n", ret); - goto err_cpvdd; + goto err_dbvdd3; } if (wm5100->pdata.ldo_ena) { @@ -2433,6 +2507,10 @@ err_ldo: err_enable: regulator_bulk_disable(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); +err_dbvdd3: + regulator_put(wm5100->dbvdd3); +err_dbvdd2: + regulator_put(wm5100->dbvdd2); err_cpvdd: regulator_put(wm5100->cpvdd); err_core: @@ -2459,6 +2537,8 @@ static int wm5100_remove(struct snd_soc_codec *codec) gpio_set_value_cansleep(wm5100->pdata.ldo_ena, 0); gpio_free(wm5100->pdata.ldo_ena); } + regulator_put(wm5100->dbvdd3); + regulator_put(wm5100->dbvdd2); regulator_put(wm5100->cpvdd); regulator_bulk_free(ARRAY_SIZE(wm5100->core_supplies), wm5100->core_supplies); -- 1.7.6.3

1 0

[alsa-devel] [PATCH 0/3] ASoC: tpa6130a2: model handling cleanup
by Peter Ujfalusi 21 Sep '11

21 Sep '11

Hello, Small cleanup for the tpa6130a2 driver for model handling: Remove the model_id from platform_data, and use the device name/device_data to distinguish between the supported models of TPA. Regards, Peter --- Peter Ujfalusi (3): ASoC: tpa6130a2: Model support cleanup OMAP: RX51: No need to configure the tpa_model ASoC: tpa6130a2: Remove model_id from platform data arch/arm/mach-omap2/board-rx51-peripherals.c | 1 - include/sound/tpa6130a2-plat.h | 6 ------ sound/soc/codecs/tpa6130a2.c | 12 +++++++++--- 3 files changed, 9 insertions(+), 10 deletions(-) -- 1.7.6.1

4 8

Re: [alsa-devel] [PATCH] ALSA: snd-aloop: make preallocated buffer size configurable
by Takashi Iwai 21 Sep '11

21 Sep '11

At Wed, 21 Sep 2011 08:26:27 -0500, Pierre-Louis Bossart wrote: > > > > Default of 2MB of physically-contiguous memory can be > > > too high for embedded solutions, make the size configurable. > > > > Hm, maybe better to have an option for global limit of preallocation? > > The problem is not necessarily specific to snd-aloop. > > Not sure I understand the comment. This code is essentially the same as the > configuration option for HDAudio, and the parameter value defines the amount > of preallocated memory, with a limit to 32MB. I meant to have a kconfig option to limit the upper-bound of pre-alloc buffer sizes for all drivers. That is, a patch like below (untested) Takashi --- diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 475455c..525ed28 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -204,6 +204,15 @@ config SND_PCM_XRUN_DEBUG sound clicking when system is loaded, it may help to determine the process or driver which causes the scheduling gaps. +config SND_PCM_MAX_PREALLOC_SIZE + int "Maximum pre-allocated buffer size" + range 0 32768 + default 0 + depends on EMBEDDED + help + Specifies the default upper-limit of PCM pre-allocation + buffer sizes in kB. Zero means unlimited. + config SND_VMASTER bool diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index 150cb7e..a09425c 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -270,6 +270,9 @@ int snd_pcm_lib_preallocate_pages(struct snd_pcm_substream *substream, int type, struct device *data, size_t size, size_t max) { + if (CONFIG_SND_PCM_MAX_PREALLOC_SIZE && + max > CONFIG_PCM_SND_MAX_PREALLOC_SIZE * 1024) + max = CONFIG_SND_PCM_PREALLOC_SIZE * 1024; substream->dma_buffer.dev.type = type; substream->dma_buffer.dev.dev = data; return snd_pcm_lib_preallocate_pages1(substream, size, max);

2 1

[alsa-devel] [RFC v2 PATCH 1/1] ASoC: soc-core: symmetry checking for each DAIs separately
by Dong Aisheng 21 Sep '11

21 Sep '11

The orginal code does not cover the case that one DAI such as codec may be shared between other two DAIs(CPU). When do symmetry checking, altough the codec DAI requires symmetry, the two CPU DAIs may still be configured to run on different rates. We change to check each DAI's state separately instead of only checking the dai link to prevent this issue. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Lars-Peter Clausen <lars(a)metafoo.de> --- Change since v1: Address the comments from Lars-Peter Clausen. * add the dai as an parameter for soc_pcm_apply_symmetry Thanks for the suggestion. --- include/sound/soc-dai.h | 3 +++ include/sound/soc.h | 2 -- sound/soc/soc-pcm.c | 40 +++++++++++++++++++++++++--------------- 3 files changed, 28 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 5ad5f3a..12d98b4 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -242,6 +242,9 @@ struct snd_soc_dai { void *playback_dma_data; void *capture_dma_data; + /* Symmetry data - only valid if symmetry is being enforced */ + unsigned int rate; + /* parent platform/codec */ union { struct snd_soc_platform *platform; diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658e..5449139 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -849,8 +849,6 @@ struct snd_soc_pcm_runtime { unsigned int complete:1; unsigned int dev_registered:1; - /* Symmetry data - only valid if symmetry is being enforced */ - unsigned int rate; long pmdown_time; /* runtime devices */ diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1aee9fc..8eb0f07 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -27,15 +27,13 @@ #include <sound/soc.h> #include <sound/initval.h> -static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream, + struct snd_soc_dai *soc_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - if (!codec_dai->driver->symmetric_rates && - !cpu_dai->driver->symmetric_rates && + if (!soc_dai->driver->symmetric_rates && !rtd->dai_link->symmetric_rates) return 0; @@ -43,19 +41,19 @@ static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) * the second can need to get its constraints before the first has * picked a rate. Complain and allow the application to carry on. */ - if (!rtd->rate) { - dev_warn(&rtd->dev, + if (!soc_dai->rate) { + dev_warn(soc_dai->dev, "Not enforcing symmetric_rates due to race\n"); return 0; } - dev_dbg(&rtd->dev, "Symmetry forces %dHz rate\n", rtd->rate); + dev_dbg(soc_dai->dev, "Symmetry forces %dHz rate\n", soc_dai->rate); ret = snd_pcm_hw_constraint_minmax(substream->runtime, SNDRV_PCM_HW_PARAM_RATE, - rtd->rate, rtd->rate); + soc_dai->rate, soc_dai->rate); if (ret < 0) { - dev_err(&rtd->dev, + dev_err(soc_dai->dev, "Unable to apply rate symmetry constraint: %d\n", ret); return ret; } @@ -185,8 +183,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) } /* Symmetry only applies if we've already got an active stream. */ - if (cpu_dai->active || codec_dai->active) { - ret = soc_pcm_apply_symmetry(substream); + if (cpu_dai->active) { + ret = soc_pcm_apply_symmetry(substream, cpu_dai); + if (ret != 0) + goto config_err; + } + + if (codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream, codec_dai); if (ret != 0) goto config_err; } @@ -288,8 +292,12 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) codec_dai->active--; codec->active--; - if (!cpu_dai->active && !codec_dai->active) - rtd->rate = 0; + /* clear the corresponding DAIs rate when inactive */ + if (!cpu_dai->active) + cpu_dai->rate = 0; + + if (!codec_dai->active) + codec_dai->rate = 0; /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. @@ -447,7 +455,9 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } - rtd->rate = params_rate(params); + /* store the rate for each DAIs */ + cpu_dai->rate = params_rate(params); + codec_dai->rate = params_rate(params); out: mutex_unlock(&rtd->pcm_mutex); -- 1.7.0.4

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[alsa-devel] ASoC/DAPM: How to make the DAPM routing complete?
by Bas Vermeulen 21 Sep '11

21 Sep '11

I am using a slightly modified version of the 88pm860x-codec.c driver from 3.x in an android modified 2.6.35 kernel. I have had to make some changes to allow both capture and playback work on the same (I2S) interface, but so far things are working. I am however having some problems getting the DAPM framework to set the correct switches needed to get the complete route to actually start. What would be needed to get this working as it should? I can post or send the driver on request, but basically the same question rises for the current driver in the 3.x kernel. I want/need to go from HS1/HS2 to DIN for playback, and from MIC1 to DOUT for capture. In order to do that, I have to set the following switches: amixer -q sset 'DAC HS1 Mux' 'Right' amixer -q sset 'DAC HS2 Mux' 'Left' amixer -q sset 'I2S Mic Mux' 'ADC' amixer -q sset 'ADC Left Mux' 'ADCL' I would like to make this automatic, but I am unsure how to express this in my dapm routing table. Note that in order to be able to use playback and capture independently on the I2S interface (within reason, it is not possible to use a different rate for instance), I had to add a SUPPLY control to the dapm widgets that turns on the I2S clock when needed. Any help, hints or pointers to the right manual to read would be appreciated. Bas Vermeulen

2 6

Re: [alsa-devel] [PATCH] ASoC: Add driver for rt5631
by Mark Brown 21 Sep '11

21 Sep '11

On Wed, Sep 07, 2011 at 11:16:35AM +0800, johnnyhsu(a)realtek.com wrote: > From d291cbcdd5c1faebd4c37b4564b33c322df51987 Mon Sep 17 00:00:00 2001 > From: Johnny Hsu <johnnyhsu(a)realtek.com> > Date: Wed, 7 Sep 2011 11:05:19 +0800 > Subject: [PATCH] ASoC: Add driver for rt5631 Applied, thanks. It'd be really helpful if where things aren't clear you could reply to code reviews - now that it's clear what the index register stuff is doing I suspect the sysfs stuff would be OK though it ought to go in debugfs not sysfs.

1 0

Re: [alsa-devel] [PATCH 3/4] ASoC: Staticise jz4740_pcm_new()
by Lars-Peter Clausen 21 Sep '11

21 Sep '11

On 09/21/2011 08:41 AM, Axel Lin wrote: > It is not used outside this driver so no need to make the symbol global. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Acked-by: Lars-Peter Clausen <lars(a)metafoo.de> Thanks. > --- > sound/soc/jz4740/jz4740-pcm.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c > index a7c9578..d1989cd 100644 > --- a/sound/soc/jz4740/jz4740-pcm.c > +++ b/sound/soc/jz4740/jz4740-pcm.c > @@ -299,7 +299,7 @@ static void jz4740_pcm_free(struct snd_pcm *pcm) > > static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); > > -int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) > +static int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > struct snd_soc_dai *dai = rtd->cpu_dai;

1 0

Re: [alsa-devel] [PATCH 1/4] ASoC: Staticise bf5xx_pcm_ac97_new()
by Mark Brown 21 Sep '11

21 Sep '11

On Wed, Sep 21, 2011 at 02:40:05PM +0800, Axel Lin wrote: > It is not used outside this driver so no need to make the symbol global. All applied, thanks.

1 0

[alsa-devel] Using the timer
by Ilya Dmitrichenko 21 Sep '11

21 Sep '11

Hi, I would like to utilise the hardware timer which is in the soundcard in an application. I suppose that I can somehow set snd-seq-timer to use a particular device as master and access that via sequencer API? The particular device that I have is Hammerfall HDSP (snd-hdsp). Cheers, -- Ilya

2 1

[alsa-devel] [PATCH 00/17] ASoC: twl6040: Cleanups and fixes
by Peter Ujfalusi 21 Sep '11

21 Sep '11

Hello, The first four patch of this series (the MFD/Input patches) has been already sent to upstream to the relevant lists, but I wanted to include here as well, since I have dependency on that series later on. If it is OK with you, it would be best to merge these MFD/Input changes through the audio tree for 3.2. Code cleanup, and fixes for the twl6040 codec driver. Main changes: - reg_cache has been rested to chip defaults - codec initialization changes - Correction for the Earphone path - AUX L/R output support - Headset/Handsfree control name changes - Headset DC offset workaround Regards, Peter I have excluded the linux-input list (but kept Dmitry) - to not spam them. --- Peter Ujfalusi (17): MFD: twl6040: Remove global pointer for platform_device MFD: twl6040: Add accessor for revision ID Input: twl6040-vibra: Use accessor to get revision information MFD: twl6040: Fix power on GPIO handling ASoC: twl6040: Chip initialization cleanup ASoC: twl6040: Use chip defaults in the initial reg_cache ASoC: twl6040: Lower the initial gains ASoC: twl6040: Fix comments for register names ASoC: twl6040: Remove strings "NULL" from DAPM route ASoC: twl6040: Introduce SW only shadow register ASoC: twl6040: Earphone path correction ASoC: twl6040: Use consistent names for Handsfree path ASoC: twl6040: Use consistent names for Headset path ASoC: twl6040: Support for AUX L/R output ASoC: twl6040/sdp4430: Change legacy DAI name ASoC/MFD: twl6040: Combine bit definitions for Headset control registers ASoC: twl6040: Workaround for headset DC offset caused pop noise drivers/input/misc/twl6040-vibra.c | 2 +- drivers/mfd/twl6040-core.c | 38 ++-- include/linux/mfd/twl6040.h | 22 +-- sound/soc/codecs/twl6040.c | 367 +++++++++++++++++++----------------- sound/soc/omap/sdp4430.c | 2 +- 5 files changed, 218 insertions(+), 213 deletions(-) -- 1.7.6.1

3 36

[alsa-devel] [PATCH] ALSA: snd-aloop: make preallocated buffer size configurable
by Pierre-Louis Bossart 21 Sep '11

21 Sep '11

Default of 2MB of physically-contiguous memory can be too high for embedded solutions, make the size configurable. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/drivers/Kconfig | 15 +++++++++++++++ sound/drivers/aloop.c | 6 +++--- 2 files changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index c896116..ad51067 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -93,6 +93,21 @@ config SND_ALOOP To compile this driver as a module, choose M here: the module will be called snd-aloop. +if SND_ALOOP +config SND_ALOOP_PREALLOC_SIZE + int "Pre-allocated buffer size for aloop driver" + range 0 32768 + default 2048 + help + Specifies the default pre-allocated buffer-size in kB for the + snd-aloop driver. The default 2048 is chosen just + for compatibility reasons. Some distributions may want to + use a smaller size when only one stream is used. + + Note that the pre-allocation size can be changed dynamically + via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. +endif + config SND_VIRMIDI tristate "Virtual MIDI soundcard" depends on SND_SEQUENCER diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index a0da775..f945573 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -556,11 +556,11 @@ static struct snd_pcm_hardware loopback_pcm_hardware = .rate_max = 192000, .channels_min = 1, .channels_max = 32, - .buffer_bytes_max = 2 * 1024 * 1024, + .buffer_bytes_max = CONFIG_SND_ALOOP_PREALLOC_SIZE * 1024, .period_bytes_min = 64, /* note check overflow in frac_pos() using pcm_rate_shift before changing period_bytes_max value */ - .period_bytes_max = 1024 * 1024, + .period_bytes_max = CONFIG_SND_ALOOP_PREALLOC_SIZE * 1024, .periods_min = 1, .periods_max = 1024, .fifo_size = 0, @@ -774,7 +774,7 @@ static int __devinit loopback_pcm_new(struct loopback *loopback, snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), - 0, 2 * 1024 * 1024); + 0, CONFIG_SND_ALOOP_PREALLOC_SIZE * 1024); return 0; } -- 1.7.6

2 2

[alsa-devel] link between HDMI ELD and PCM devices
by Pierre-Louis Bossart 21 Sep '11

21 Sep '11

Hi, I've been trying for some time to expose the HDMI ELD information in a new control. The problem I am facing with the HDMI/HDA code is that the ELD is linked to an output pin, and I can't find an elegant way to link this output pin with a PCM device number to be used by my control. For a single output, I could hard-code something but it would probably not work with multiple monitors handled by recent hardware. There is some existing code such as pin_nid_to_pin_index() in patch_hdmi.c, but I can't figure out what this pin_idx is, and how to find the related PCM device number. Any suggestions? Thanks, -Pierre

4 5

Re: [alsa-devel] Questions for dummy codec driver
by Mark Brown 21 Sep '11

21 Sep '11

On Wed, Sep 21, 2011 at 10:23:18AM +0800, Leo Yan wrote: > I have a question for the dummy codec. Don't start new threads by randomly replying to unrelated mails. > As you know, sometimes maybe we only need set the timing from the > processor, and codec just only need tx/rx data; So there do not need > the codec driver. Even a dumb CODEC will have some constraints on things like the sample rates it can support which need to be conveyed to the framework and ideally the audio routing out of the CODEC should be visible for integration with any external output drivers or similar. > After browser the code, there have NO such a dummy codec driver; and > for soc-core, there have not such flags to support this situation. > Only what i get is sound/soc/codecs/spdif_transciever.c, but it's > specific for TI's DaVinci platform. Not at all, what makes you believe that it is platform specific?

1 0

[alsa-devel] [PATCH] ASoC: Ensure we generate a driver name
by Mark Brown 21 Sep '11

21 Sep '11

Commit 873bd4c (ASoC: Don't set invalid name string to snd_card->driver field) broke generation of a driver name for all ASoC cards relying on the automatic generation of one. Fix this by using the old default with spaces replaced by underscores. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 18 +++++++++++++++--- 1 files changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 10e5cde..bd20154 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -30,6 +30,7 @@ #include <linux/bitops.h> #include <linux/debugfs.h> #include <linux/platform_device.h> +#include <linux/ctype.h> #include <linux/slab.h> #include <sound/ac97_codec.h> #include <sound/core.h> @@ -1432,9 +1433,20 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); - if (card->driver_name) - strlcpy(card->snd_card->driver, card->driver_name, - sizeof(card->snd_card->driver)); + snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), + "%s", card->driver_name ? card->driver_name : card->name); + for (i = 0; i < ARRAY_SIZE(card->snd_card->driver); i++) { + switch (card->snd_card->driver[i]) { + case '_': + case '-': + case '\0': + break; + default: + if (!isalnum(card->snd_card->driver[i])) + card->snd_card->driver[i] = '_'; + break; + } + } if (card->late_probe) { ret = card->late_probe(card); -- 1.7.6.3

2 2

Re: [alsa-devel] [PATCH] ASoC: Flush Samsung DMA on free
by Kukjin Kim 21 Sep '11

21 Sep '11

Boojin Kim wrote: > > > Mark Brown wrote: > > > > Ever since it was written the Samsung DMA driver has had a TODO in the > > hw_free() function wondering if we need to flush the DMA buffers. Up > > until > > now the answer has been no but with the recent improvements Boojin has > > done to the DMA infrastructure for the Samsung port the answer has > > changed > > to yes for at least S3C6410 systems. > > > > If we don't then when we next prepare() the channel the API will get > > confused trying to run callbacks on the transfers hanging around from > > the > > previous time the stream was open and oops. > > > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > > > Your patch give good effect to all Samsung SoCs. > Thanks a lot. > > Acked-by: Boojin Kim <boojin.kim(a)samsung.com> > > > --- > > > > Kukjin, this should probably get added to your branch assuming it's OK > > for other CPUs. > > Looks ok to me, will apply. Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd. > > sound/soc/samsung/dma.c | 2 +- > > 1 files changed, 1 insertions(+), 1 deletions(-) > > > > diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c > > index 851346f..2d622b6 100644 > > --- a/sound/soc/samsung/dma.c > > +++ b/sound/soc/samsung/dma.c > > @@ -198,10 +198,10 @@ static int dma_hw_free(struct snd_pcm_substream > > *substream) > > > > pr_debug("Entered %s\n", __func__); > > > > - /* TODO - do we need to ensure DMA flushed */ > > snd_pcm_set_runtime_buffer(substream, NULL); > > > > if (prtd->params) { > > + prtd->params->ops->flush(prtd->params->ch); > > prtd->params->ops->release(prtd->params->ch, > > prtd->params->client); > > prtd->params = NULL; > > -- > > 1.7.6.3

1 0

[alsa-devel] [PATCH] ALSA: hda: hdmi: Hint matching between input devices and pcm devices
by David Henningsson 21 Sep '11

21 Sep '11

Hi Takashi, You went on vacation when this patch was posted and discussed, but would you mind reviewing/committing this patch now that you're back? It was reviewed by Stephen Warren who thought the patch was okay, but had some thoughts on doing other changes to the format of that string at the same time [1]. I had to distro patch a 3.0 adapted version of this patch into Ubuntu in order to meet the Ubuntu 11.10 kernel deadline, so for that reason I'd say that the ",pcm=x" format would be preferrable for me, and while I don't mind further improvements to the string format in the future, this would be a step in the right direction and IMO this patch should not block on such future improvements. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-August/043258.html

2 1

Re: [alsa-devel] [PATCH 4/4] ASoC: Staticise nuc900_dma_getposition()
by Wan ZongShun 21 Sep '11

21 Sep '11

2011/9/21 Axel Lin <axel.lin(a)gmail.com>: > It is not used outside this driver so no need to make the symbol global. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/nuc900/nuc900-pcm.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c > index e46d551..4e3626b 100644 > --- a/sound/soc/nuc900/nuc900-pcm.c > +++ b/sound/soc/nuc900/nuc900-pcm.c > @@ -227,7 +227,7 @@ static int nuc900_dma_trigger(struct snd_pcm_substream *substream, int cmd) > return ret; > } > > -int nuc900_dma_getposition(struct snd_pcm_substream *substream, > +static int nuc900_dma_getposition(struct snd_pcm_substream *substream, > dma_addr_t *src, dma_addr_t *dst) > { > struct snd_pcm_runtime *runtime = substream->runtime; > -- > 1.7.4.1 > thanks! Acked- by: Wan Zongshun <mcuos.com(a)gmail.com> > > > -- Wan ZongShun. www.mcuos.com

1 0

[alsa-devel] [RFC] Remove non-exist device and control from snd-ymfpci
by Raymond Yau 21 Sep '11

21 Sep '11

My ymf724f does not have spdif, only with ac97 stac9704 Is it feasible to remove playback device 2 (rear) and control (e.g. 4 ch Duplication) by checking sdac bit of ac97 registers to know that it does not support 4 channels ? How about device 1 spdif since this card does not have spdif connector ? what is the difference of capture device 0 and device 3 ? !!PCI Soundcards installed in the system !!-------------------------------------- 04:02.0 Multimedia audio controller: Yamaha Corporation YMF-724F [DS-1 Audio Controller] (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 04:02.0 0401: 1073:000d (rev 03) Subsystem: 1073:000d Simple mixer control 'IEC958',0 Capabilities: cvolume pswitch pswitch-joined cswitch cswitch-joined penum Playback channels: Mono Capture channels: Front Left - Front Right Limits: Capture 0 - 16383 Mono: Playback [off] Front Left: Capture 0 [0%] [-99999.99dB] [off] Front Right: Capture 0 [0%] [-99999.99dB] [off] Simple mixer control 'IEC958 AC97',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 16383 Mono: Front Left: Playback 16383 [100%] [0.00dB] Front Right: Playback 16383 [100%] [0.00dB] Simple mixer control 'IEC958 AC97',1 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 16383 Mono: Front Left: Playback 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Simple mixer control 'IEC958 Loop',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'IEC958',1 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 16383 Front Left: Capture 0 [0%] [-99999.99dB] Front Right: Capture 0 [0%] [-99999.99dB] Simple mixer control '4ch Duplication',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'ADC',0 Capabilities: pvolume cvolume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 16383 Capture 0 - 16383 Front Left: Playback 0 [0%] [-99999.99dB] Capture 16383 [100%] [0.00dB] Front Right: Playback 0 [0%] [-99999.99dB] Capture 16383 [100%] [0.00dB] Simple mixer control 'ADC',1 Capabilities: pvolume cvolume penum Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 16383 Capture 0 - 16383 Front Left: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Simple mixer control 'Digital',0 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 16383 Front Left: Capture 16383 [100%] [0.00dB] Front Right: Capture 16383 [100%] [0.00dB] Simple mixer control 'Digital',1 Capabilities: cvolume penum Capture channels: Front Left - Front Right Limits: Capture 0 - 16383 Front Left: Capture 16383 [100%] [0.00dB] Front Right: Capture 16383 [100%] [0.00dB] **** List of PLAYBACK Hardware Devices **** card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 0: YMFPCI [YMFPCI] Subdevices: 32/32 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 Subdevice #8: subdevice #8 Subdevice #9: subdevice #9 Subdevice #10: subdevice #10 Subdevice #11: subdevice #11 Subdevice #12: subdevice #12 Subdevice #13: subdevice #13 Subdevice #14: subdevice #14 Subdevice #15: subdevice #15 Subdevice #16: subdevice #16 Subdevice #17: subdevice #17 Subdevice #18: subdevice #18 Subdevice #19: subdevice #19 Subdevice #20: subdevice #20 Subdevice #21: subdevice #21 Subdevice #22: subdevice #22 Subdevice #23: subdevice #23 Subdevice #24: subdevice #24 Subdevice #25: subdevice #25 Subdevice #26: subdevice #26 Subdevice #27: subdevice #27 Subdevice #28: subdevice #28 Subdevice #29: subdevice #29 Subdevice #30: subdevice #30 Subdevice #31: subdevice #31 card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 1: YMFPCI - IEC958 [YMFPCI - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 2: YMFPCI - Rear [YMFPCI - Rear PCM] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 0: YMFPCI [YMFPCI] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: YMF724F [Yamaha DS-1 (YMF724F)], device 3: YMFPCI - PCM2 [YMFPCI - AC'97] Subdevices: 1/1 Subdevice #0: subdevice #0 !!AC97 Codec information !!--------------------------- 0-0/0: SigmaTel STAC9704 PCI Subsys Vendor: 0x0000 PCI Subsys Device: 0x0000 Flags: 0 Capabilities : DAC resolution : 18-bit ADC resolution : 18-bit 3D enhancement : SigmaTel 3D Enhancement Current setup Mic gain : +0dB [+0dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off 0:00 = 6940 0:02 = 1313 0:04 = 8000 0:06 = 801f 0:08 = ffff 0:0a = 9f1f 0:0c = 801f 0:0e = 801f 0:10 = 9f1f 0:12 = 9f1f 0:14 = 9f1f 0:16 = 9f1f 0:18 = 0808 0:1a = 9818 0:1c = 0808 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 000f 0:28 = ffff 0:2a = ffff 0:2c = ffff 0:2e = ffff 0:30 = ffff 0:32 = ffff 0:34 = ffff 0:36 = ffff 0:38 = ffff 0:3a = ffff 0:3c = 0000 0:3e = 0000 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0000 0:4e = 0000 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 0000 0:5e = 0000 0:60 = 0000 0:62 = 0000 0:64 = 0000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 0000 0:6e = 0000 0:70 = ffff 0:72 = ffff 0:74 = ffff 0:76 = ffff 0:78 = ffff 0:7a = ffff 0:7c = 8384 0:7e = 7605

2 2

[alsa-devel] [RFC PATCH] ALSA: hdmi: expose ELD control
by Pierre-Louis Bossart 21 Sep '11

21 Sep '11

Applications may want to read ELD information to understand what codecs are supported on the HDMI receiver and handle the a-v delay for better lip-sync. ELD information is exposed in a BYTES control named "ELD <device name>". Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 9 ++-- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_hdmi.c | 91 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 97 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 28ce17d..8f16b20 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -335,21 +335,20 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); size = 128; } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE || size > PAGE_SIZE) { snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); return -ERANGE; } - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; + /* update info */ + eld->eld_size = size; + buf = eld->eld_buffer; for (i = 0; i < size; i++) buf[i] = hdmi_get_eld_byte(codec, nid, i); ret = hdmi_update_eld(eld, buf, size); - kfree(buf); return ret; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 9ed4b0d..cd579a6 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -609,6 +609,7 @@ struct cea_sad { }; #define ELD_FIXED_BYTES 20 +#define ELD_MAX_SIZE 256 #define ELD_MAX_MNL 16 #define ELD_MAX_SAD 16 @@ -633,6 +634,7 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19cb72d..a5b7245 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -324,6 +324,89 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) return -EINVAL; } +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_nid; + int pin_idx; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + + pin_nid = kcontrol->private_value; + spec = codec->spec; + + pin_idx = pin_nid_to_pin_index(spec, pin_nid); + if (pin_idx < 0) { + uinfo->count = 0; + } else { + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + uinfo->count = eld->eld_size; + } + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_nid; + int pin_idx; + + spec = codec->spec; + pin_nid = kcontrol->private_value; + + pin_idx = pin_nid_to_pin_index(spec, pin_nid); + if (pin_idx < 0) { + memset(ucontrol->value.bytes.data, 0, ELD_MAX_SIZE); + } else { + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, + (eld->eld_size > ELD_MAX_SIZE) ? + ELD_MAX_SIZE : eld->eld_size); + } + return 0; +} + +static struct snd_kcontrol_new eld_bytes_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ELD ", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, +}; + +static int hdmi_create_eld_bytes_ctl(struct hda_codec *codec, hda_nid_t nid, + char *name) +{ + struct snd_kcontrol *kctl; + struct hdmi_spec *spec = codec->spec; + int pin_idx; + int err; + + pin_idx = pin_nid_to_pin_index(spec, nid); + if (pin_idx < 0) + return -EINVAL; + + kctl = snd_ctl_new1(&eld_bytes_ctl, codec); + if (!kctl) + return -ENOMEM; + err = snd_hda_ctl_add(codec, nid, kctl); + if (err < 0) + return err; + kctl->private_value = nid; + strcat(kctl->id.name, name); + + return 0; +} + #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) @@ -1145,6 +1228,7 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { struct hda_pcm *info; struct hda_pcm_stream *pstr; + int ret; info = &spec->pcm_rec[pin_idx]; info->name = generic_hdmi_pcm_names[pin_idx]; @@ -1154,6 +1238,13 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) pstr->substreams = 1; pstr->ops = generic_ops; /* other pstr fields are set in open */ + + /* add control for ELD Bytes */ + ret = hdmi_create_eld_bytes_ctl(codec, + spec->pins[pin_idx].pin_nid, + info->name); + if (ret < 0) + return ret; } codec->num_pcms = spec->num_pins; -- 1.7.6.2

1 0

[PATCH] ALSA: hdmi: expose ELD control
by Pierre-Louis Bossart 21 Sep '11

21 Sep '11

Applications may want to read ELD information to understand what codecs are supported on the HDMI receiver and handle the a-v delay for better lip-sync. ELD information is exposed in a device-specific IFACE_PCM kcontrol. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/hda/hda_eld.c | 9 ++--- sound/pci/hda/hda_local.h | 2 + sound/pci/hda/patch_hdmi.c | 85 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 91 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index 28ce17d..8f16b20 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -335,21 +335,20 @@ int snd_hdmi_get_eld(struct hdmi_eld *eld, snd_printd(KERN_INFO "HDMI: ELD buf size is 0, force 128\n"); size = 128; } - if (size < ELD_FIXED_BYTES || size > PAGE_SIZE) { + if (size < ELD_FIXED_BYTES || size > ELD_MAX_SIZE || size > PAGE_SIZE) { snd_printd(KERN_INFO "HDMI: invalid ELD buf size %d\n", size); return -ERANGE; } - buf = kmalloc(size, GFP_KERNEL); - if (!buf) - return -ENOMEM; + /* update info */ + eld->eld_size = size; + buf = eld->eld_buffer; for (i = 0; i < size; i++) buf[i] = hdmi_get_eld_byte(codec, nid, i); ret = hdmi_update_eld(eld, buf, size); - kfree(buf); return ret; } diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 9ed4b0d..cd579a6 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -609,6 +609,7 @@ struct cea_sad { }; #define ELD_FIXED_BYTES 20 +#define ELD_MAX_SIZE 256 #define ELD_MAX_MNL 16 #define ELD_MAX_SAD 16 @@ -633,6 +634,7 @@ struct hdmi_eld { int spk_alloc; int sad_count; struct cea_sad sad[ELD_MAX_SAD]; + char eld_buffer[ELD_MAX_SIZE]; #ifdef CONFIG_PROC_FS struct snd_info_entry *proc_entry; #endif diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 19cb72d..5ea9b55 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -324,6 +324,83 @@ static int cvt_nid_to_cvt_index(struct hdmi_spec *spec, hda_nid_t cvt_nid) return -EINVAL; } +static int hdmi_eld_ctl_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + pin_idx = kcontrol->private_value; + spec = codec->spec; + + if (pin_idx < 0) { + uinfo->count = 0; + } else { + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + uinfo->count = eld->eld_size; + } + return 0; +} + +static int hdmi_eld_ctl_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct hdmi_spec *spec; + struct hdmi_spec_per_pin *per_pin; + struct hdmi_eld *eld; + int pin_idx; + + spec = codec->spec; + pin_idx = kcontrol->private_value; + + if (pin_idx < 0) { + memset(ucontrol->value.bytes.data, 0, ELD_MAX_SIZE); + } else { + per_pin = &spec->pins[pin_idx]; + eld = &per_pin->sink_eld; + memcpy(ucontrol->value.bytes.data, eld->eld_buffer, + (eld->eld_size > ELD_MAX_SIZE) ? + ELD_MAX_SIZE : eld->eld_size); + } + return 0; +} + +static struct snd_kcontrol_new eld_bytes_ctl = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "ELD", + .info = hdmi_eld_ctl_info, + .get = hdmi_eld_ctl_get, +}; + +static int hdmi_create_eld_ctl(struct hda_codec *codec, int pin_idx, + int device) +{ + struct snd_kcontrol *kctl; + struct hdmi_spec *spec = codec->spec; + int err; + + if (pin_idx < 0) + return -EINVAL; + + kctl = snd_ctl_new1(&eld_bytes_ctl, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = pin_idx; + kctl->id.device = device; + + err = snd_hda_ctl_add(codec, spec->pins[pin_idx].pin_nid, kctl); + if (err < 0) + return err; + + return 0; +} + #ifdef BE_PARANOID static void hdmi_get_dip_index(struct hda_codec *codec, hda_nid_t pin_nid, int *packet_index, int *byte_index) @@ -1176,6 +1253,14 @@ static int generic_hdmi_build_controls(struct hda_codec *codec) if (err < 0) return err; snd_hda_spdif_ctls_unassign(codec, pin_idx); + + /* add control for ELD Bytes */ + err = hdmi_create_eld_ctl(codec, + pin_idx, + spec->pcm_rec[pin_idx].device); + + if (err < 0) + return err; } return 0; -- 1.7.6.2 --=-TOAARVHGJMGbcf5NX2ZN--

1 0

[alsa-devel] [PATCH] ASoC: Factor out per-widget DAPM power checks
by Mark Brown 21 Sep '11

21 Sep '11

The indentation is getting a little deep. Should be straight code motion, no functional changes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 112 +++++++++++++++++++++++++++----------------------- 1 files changed, 60 insertions(+), 52 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6a1e13e..84d1d79 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1191,6 +1191,65 @@ static void dapm_post_sequence_async(void *data, async_cookie_t cookie) } } +static void dapm_power_one_widget(struct snd_soc_dapm_widget *w, + struct list_head *up_list, + struct list_head *down_list) +{ + struct snd_soc_dapm_context *d; + int power; + + switch (w->id) { + case snd_soc_dapm_pre: + dapm_seq_insert(w, down_list, false); + break; + case snd_soc_dapm_post: + dapm_seq_insert(w, up_list, true); + break; + + default: + if (!w->power_check) + break; + + if (!w->force) + power = w->power_check(w); + else + power = 1; + + if (power) { + d = w->dapm; + + /* Supplies and micbiases only bring the + * context up to STANDBY as unless something + * else is active and passing audio they + * generally don't require full power. + */ + switch (w->id) { + case snd_soc_dapm_supply: + case snd_soc_dapm_micbias: + if (d->target_bias_level < SND_SOC_BIAS_STANDBY) + d->target_bias_level = SND_SOC_BIAS_STANDBY; + break; + default: + d->target_bias_level = SND_SOC_BIAS_ON; + break; + } + } + + if (w->power == power) + break; + + trace_snd_soc_dapm_widget_power(w, power); + + if (power) + dapm_seq_insert(w, up_list, true); + else + dapm_seq_insert(w, down_list, false); + + w->power = power; + break; + } +} + /* * Scan each dapm widget for complete audio path. * A complete path is a route that has valid endpoints i.e.:- @@ -1209,7 +1268,6 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) LIST_HEAD(down_list); LIST_HEAD(async_domain); enum snd_soc_bias_level bias; - int power; trace_snd_soc_dapm_start(card); @@ -1228,57 +1286,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) * lists indicating if they should be powered up or down. */ list_for_each_entry(w, &card->widgets, list) { - switch (w->id) { - case snd_soc_dapm_pre: - dapm_seq_insert(w, &down_list, false); - break; - case snd_soc_dapm_post: - dapm_seq_insert(w, &up_list, true); - break; - - default: - if (!w->power_check) - continue; - - if (!w->force) - power = w->power_check(w); - else - power = 1; - - if (power) { - d = w->dapm; - - /* Supplies and micbiases only bring - * the context up to STANDBY as unless - * something else is active and - * passing audio they generally don't - * require full power. - */ - switch (w->id) { - case snd_soc_dapm_supply: - case snd_soc_dapm_micbias: - if (d->target_bias_level < SND_SOC_BIAS_STANDBY) - d->target_bias_level = SND_SOC_BIAS_STANDBY; - break; - default: - d->target_bias_level = SND_SOC_BIAS_ON; - break; - } - } - - if (w->power == power) - continue; - - trace_snd_soc_dapm_widget_power(w, power); - - if (power) - dapm_seq_insert(w, &up_list, true); - else - dapm_seq_insert(w, &down_list, false); - - w->power = power; - break; - } + dapm_power_one_widget(w, &up_list, &down_list); } /* If there are no DAPM widgets then try to figure out power from the -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ASoC: Trace and collect statistics for DAPM graph walking
by Mark Brown 20 Sep '11

20 Sep '11

One of the longest standing areas for improvement in ASoC has been the DAPM algorithm - it repeats the same checks many times whenever it is run and makes no effort to limit the areas of the graph it checks meaning we do an awful lot of walks over the full graph. This has never mattered too much as the size of the graph has generally been small in relation to the size of the devices supported and the speed of CPUs but it is annoying. In preparation for work on improving this insert a trace point after the graph walk has been done. This gives us specific timing information for the walk, and in order to give quantifiable (non-benchmark) numbers also count every time we check a link or check the power for a widget and report those numbers. Substantial changes in the algorithm may require tweaks to the stats but they should be useful for simpler things. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++++ include/sound/soc.h | 1 + include/trace/events/asoc.h | 22 ++++++++++++++++++++++ sound/soc/soc-dapm.c | 17 +++++++++++++++++ 4 files changed, 45 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 350b1b3..0e2d017 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -537,4 +537,9 @@ struct snd_soc_dapm_widget_list { struct snd_soc_dapm_widget *widgets[0]; }; +struct snd_soc_dapm_stats { + int power_checks; + int path_checks; +}; + #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index 24e17be..006f4f6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -828,6 +828,7 @@ struct snd_soc_card { /* Generic DAPM context for the card */ struct snd_soc_dapm_context dapm; + struct snd_soc_dapm_stats dapm_stats; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_card_root; diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index 603f5a0..2e1adf6 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -216,6 +216,28 @@ DEFINE_EVENT(snd_soc_dapm_widget, snd_soc_dapm_widget_event_done, ); +TRACE_EVENT(snd_soc_dapm_walk_done, + + TP_PROTO(struct snd_soc_card *card), + + TP_ARGS(card), + + TP_STRUCT__entry( + __string( name, card->name ) + __field( int, power_checks ) + __field( int, path_checks ) + ), + + TP_fast_assign( + __assign_str(name, card->name); + __entry->power_checks = card->dapm_stats.power_checks; + __entry->path_checks = card->dapm_stats.path_checks; + ), + + TP_printk("%s: %d power checks, %d path checks", __get_str(name), + (int)__entry->power_checks, (int)__entry->path_checks) +); + TRACE_EVENT(snd_soc_jack_irq, TP_PROTO(const char *name), diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4a440b5..6a1e13e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -48,6 +48,8 @@ #include <trace/events/asoc.h> +#define DAPM_UPDATE_STAT(widget, val) widget->dapm->card->dapm_stats.val++; + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -649,6 +651,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + DAPM_UPDATE_STAT(widget, path_checks); + if (widget->id == snd_soc_dapm_supply) return 0; @@ -697,6 +701,8 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) struct snd_soc_dapm_path *path; int con = 0; + DAPM_UPDATE_STAT(widget, path_checks); + if (widget->id == snd_soc_dapm_supply) return 0; @@ -767,6 +773,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; + DAPM_UPDATE_STAT(w, power_checks); + in = is_connected_input_ep(w); dapm_clear_walk(w->dapm); out = is_connected_output_ep(w); @@ -779,6 +787,8 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) { int in; + DAPM_UPDATE_STAT(w, power_checks); + if (w->active) { in = is_connected_input_ep(w); dapm_clear_walk(w->dapm); @@ -793,6 +803,8 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) { int out; + DAPM_UPDATE_STAT(w, power_checks); + if (w->active) { out = is_connected_output_ep(w); dapm_clear_walk(w->dapm); @@ -808,6 +820,8 @@ static int dapm_supply_check_power(struct snd_soc_dapm_widget *w) struct snd_soc_dapm_path *path; int power = 0; + DAPM_UPDATE_STAT(w, power_checks); + /* Check if one of our outputs is connected */ list_for_each_entry(path, &w->sinks, list_source) { if (path->weak) @@ -1208,6 +1222,8 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) } } + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); + /* Check which widgets we need to power and store them in * lists indicating if they should be powered up or down. */ @@ -1299,6 +1315,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) list_for_each_entry(d, &card->dapm_list, list) d->target_bias_level = bias; + trace_snd_soc_dapm_walk_done(card); /* Run all the bias changes in parallel */ list_for_each_entry(d, &dapm->card->dapm_list, list) -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ALSA: HDA: Refactor Realtek's automute
by David Henningsson 20 Sep '11

20 Sep '11

Ok, my automute refactoring has been rebased to your latest tree. I've tested it on my own machine, but I don't know if wider testing is needed to ensure there are no regressions. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 5

Re: [alsa-devel] [PATCH] ASoC: fsl: Fix error handling if platform_device_add fails
by Timur Tabi 20 Sep '11

20 Sep '11

Axel Lin wrote: > Call platform_device_put() instead of platform_device_unregister() if > platform_device_add() fails. What's wrong with calling platform_device_unregister()? -- Timur Tabi Linux kernel developer at Freescale

1 0

[alsa-devel] ASoC updates for 3.2
by Mark Brown 20 Sep '11

20 Sep '11

The following changes since commit 30ab1e78864ca5781de5b1fb501bed9df2c215f1: ASoC: ad193x: Setup regmap read and write flag masks for SPI (2011-09-09 11:13:31 -0700) are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.2 Arnd Bergmann (1): ASoC: Fix trivial build regression in Kirkwood I2S Axel Lin (7): ASoC: samsung: Fix checking return value of clk_get ASoC: bf5xx-ad73311: Fix prototype for bf5xx_probe ASoC: tpa6130a2: Remove obsolete cleanup for clientdata ASoC: wm8995: Return -EINVAL if device ID mismatch ASoC: wm8995: Remove unused i2c variable in wm8995_remove() ASoC: sn95031: Staticize sn95031_pcm_hw_params ASoC: fsl: Fix error handling if platform_device_add fails Ben Gardiner (1): ASoC: davinci-pcm: trivial: replace link with actual chan/link Dong Aisheng (1): ASoC: mxs-saif: add record function Lars-Peter Clausen (1): ASoC: ssm2602: Do not dereference codec->control_data Mark Brown (13): regulator: Implement deferred disable support ASoC: Disable WM8996 CPVDD supply when not in use ASoC: Correct channel numbers for WM8996 AIF2 ASoC: Fix WM8996 DC servo operation without IRQ Merge branch 'for-3.1' into for-3.2 ASoC: Display the error code when we fail to add a DAPM control ASoC: WM8996 only needs bandgap for analogue functionality ASoC: Initial WM8996 headphone impedance measurement support ASoC: Add line loads to the list of supported detections for Speyside ASoC: Fix unused variable warning in WM8996 ASoC: Refcount WM8996 bandgap from FLL too ASoC: Remove bitrotted wm8962_resume() Merge branch 'for-3.1' into for-3.2 Mika Westerberg (7): ASoC: snappercl15: convert to use snd_soc_register_card() ARM: ep93xx: simone: register audio platform device ARM: ep93xx: edb93xx: register audio platform device ARM: ep93xx: snappercl15: register audio platform device ASoC: ep93xx-pcm: add MODULE_ALIAS ASoC: simone: convert to use snd_soc_register_card() ASoC: edb93xx: convert to use snd_soc_register_card() Peter Ujfalusi (8): MFD: twl6040: Remove global pointer for platform_device MFD: twl6040: Add accessor for revision ID Input: twl6040-vibra: Use accessor to get revision information MFD: twl6040: Fix power on GPIO handling ASoC: twl6040: Chip initialization cleanup ASoC: twl6040: Use chip defaults in the initial reg_cache ASoC: twl6040: Fix the number of channels for vibra ASoC: twl6040: Correct supported number of playback channels Timur Tabi (3): ASoC: improve asynchronous mode support in the fsl_ssi driver ASoC: support sample sizes properly in the WM8776 codec driver ASoC: support all possible sample rates in the WM8776 driver arch/arm/mach-ep93xx/edb93xx.c | 6 + arch/arm/mach-ep93xx/simone.c | 13 ++- arch/arm/mach-ep93xx/snappercl15.c | 13 ++- drivers/input/misc/twl6040-vibra.c | 2 +- drivers/mfd/twl6040-core.c | 38 +++--- drivers/regulator/core.c | 59 ++++++++ include/linux/mfd/twl6040.h | 8 +- include/linux/regulator/consumer.h | 7 + include/linux/regulator/driver.h | 3 + include/sound/saif.h | 16 ++ sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/codecs/sn95031.c | 2 +- sound/soc/codecs/ssm2602.c | 3 +- sound/soc/codecs/tpa6130a2.c | 1 - sound/soc/codecs/twl6040.c | 114 +++------------ sound/soc/codecs/wm8776.c | 38 +++--- sound/soc/codecs/wm8962.c | 26 ---- sound/soc/codecs/wm8995.c | 3 +- sound/soc/codecs/wm8996.c | 263 +++++++++++++++++++++++++++------- sound/soc/davinci/davinci-pcm.c | 123 ++++++++--------- sound/soc/ep93xx/edb93xx.c | 60 +++++---- sound/soc/ep93xx/ep93xx-pcm.c | 1 + sound/soc/ep93xx/simone.c | 64 +++++---- sound/soc/ep93xx/snappercl15.c | 53 +++++--- sound/soc/fsl/fsl_ssi.c | 145 +++++++++---------- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/mxs/mxs-saif.c | 145 +++++++++++++++++-- sound/soc/mxs/mxs-saif.h | 4 + sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 8 +- sound/soc/samsung/speyside.c | 3 +- sound/soc/soc-dapm.c | 4 +- 35 files changed, 775 insertions(+), 466 deletions(-) create mode 100644 include/sound/saif.h

2 1

[alsa-devel] [PATCH] ASoC: Ensure we generate a driver name
by Mark Brown 20 Sep '11

20 Sep '11

Commit 873bd4c (ASoC: Don't set invalid name string to snd_card->driver field) broke generation of a driver name for all ASoC cards relying on the automatic generation of one. Fix this by using the old default with spaces replaced by underscores. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 17 ++++++++++++++--- 1 files changed, 14 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 10e5cde..3096c0d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -30,6 +30,7 @@ #include <linux/bitops.h> #include <linux/debugfs.h> #include <linux/platform_device.h> +#include <linux/ctype.h> #include <linux/slab.h> #include <sound/ac97_codec.h> #include <sound/core.h> @@ -1432,9 +1433,19 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); - if (card->driver_name) - strlcpy(card->snd_card->driver, card->driver_name, - sizeof(card->snd_card->driver)); + snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), + "%s", card->driver_name ? card->driver_name : card->name); + for (i = 0; i < ARRAY_SIZE(card->snd_card->driver); i++) { + switch (card->snd_card->driver[i]) { + case '_': + case '-': + break; + default: + if (!isalnum(card->snd_card->driver[i])) + card->snd_card->driver[i] = '_'; + break; + } + } if (card->late_probe) { ret = card->late_probe(card); -- 1.7.6.3

2 5

[alsa-devel] [PATCH] ASoC: Ensure we generate a card name
by Mark Brown 20 Sep '11

20 Sep '11

Commit 873bd4c (ASoC: Don't set invalid name string to snd_card->driver field) broke generation of a driver name for all ASoC cards relying on the automatic generation of one. Fix this by using the old default with spaces replaced by underscores. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 10e5cde..7e30c08 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1432,9 +1432,11 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); - if (card->driver_name) - strlcpy(card->snd_card->driver, card->driver_name, - sizeof(card->snd_card->driver)); + snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), + "%s", card->driver_name ? card->driver_name : card->name); + for (i = 0; i < ARRAY_SIZE(card->snd_card->driver); i++) + if (card->snd_card->driver[i] == ' ') + card->snd_card->driver[i] = '_'; if (card->late_probe) { ret = card->late_probe(card); -- 1.7.6.3

2 9

[alsa-devel] [PATCH] ASoC: ssm2602: Do not dereference codec->control_data
by Lars-Peter Clausen 20 Sep '11

20 Sep '11

The driver assumes that control_data points to the drivers i2c_client struct, but this is no longer the case since the ASoC core has switched to regmap. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm2602.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 84f4ad5..cceb002 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -294,7 +294,6 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec *codec = rtd->codec; struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); - struct i2c_client *i2c = codec->control_data; struct snd_pcm_runtime *master_runtime; /* The DAI has shared clocks so if we already have a playback or @@ -303,7 +302,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream, */ if (ssm2602->master_substream) { master_runtime = ssm2602->master_substream->runtime; - dev_dbg(&i2c->dev, "Constraining to %d bits at %dHz\n", + dev_dbg(codec->dev, "Constraining to %d bits at %dHz\n", master_runtime->sample_bits, master_runtime->rate); -- 1.7.2.5

3 2

Re: [alsa-devel] [PATCH] ASoC: fsl: Fix error handling if platform_device_add fails
by Mark Brown 20 Sep '11

20 Sep '11

On Tue, Sep 20, 2011 at 03:09:00PM +0800, Axel Lin wrote: > Call platform_device_put() instead of platform_device_unregister() if > platform_device_add() fails. Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: fsl: Fix error handling if platform_device_add fails
by Liam Girdwood 20 Sep '11

20 Sep '11

On Tue, 2011-09-20 at 15:09 +0800, Axel Lin wrote: > Call platform_device_put() instead of platform_device_unregister() if > platform_device_add() fails. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/fsl/mpc8610_hpcd.c | 2 +- > sound/soc/fsl/p1022_ds.c | 2 +- > 2 files changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c > index 358f0ba..31af405 100644 > --- a/sound/soc/fsl/mpc8610_hpcd.c > +++ b/sound/soc/fsl/mpc8610_hpcd.c > @@ -505,7 +505,7 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) > return 0; > > error_sound: > - platform_device_unregister(sound_device); > + platform_device_put(sound_device); > error: > kfree(machine_data); > error_alloc: > diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c > index e8849ed..2c064a9 100644 > --- a/sound/soc/fsl/p1022_ds.c > +++ b/sound/soc/fsl/p1022_ds.c > @@ -506,7 +506,7 @@ static int p1022_ds_probe(struct platform_device *pdev) > > error: > if (sound_device) > - platform_device_unregister(sound_device); > + platform_device_put(sound_device); > > kfree(mdata); > error_put: Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

[alsa-devel] [PATCH] ASoC: Remove bitrotted wm8962_resume()
by Mark Brown 20 Sep '11

20 Sep '11

This functionality is now subsumed within the bias management, using the standard cache management functionality, without assuming the cache type. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 26 -------------------------- 1 files changed, 0 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index ae7a263..ac2a859 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3680,31 +3680,6 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) } EXPORT_SYMBOL_GPL(wm8962_mic_detect); -#ifdef CONFIG_PM -static int wm8962_resume(struct snd_soc_codec *codec) -{ - u16 *reg_cache = codec->reg_cache; - int i; - - /* Restore the registers */ - for (i = 1; i < codec->driver->reg_cache_size; i++) { - switch (i) { - case WM8962_SOFTWARE_RESET: - continue; - default: - break; - } - - if (reg_cache[i] != wm8962_reg[i]) - snd_soc_write(codec, i, reg_cache[i]); - } - - return 0; -} -#else -#define wm8962_resume NULL -#endif - #if defined(CONFIG_INPUT) || defined(CONFIG_INPUT_MODULE) static int beep_rates[] = { 500, 1000, 2000, 4000, @@ -4228,7 +4203,6 @@ static int wm8962_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8962 = { .probe = wm8962_probe, .remove = wm8962_remove, - .resume = wm8962_resume, .set_bias_level = wm8962_set_bias_level, .reg_cache_size = WM8962_MAX_REGISTER + 1, .reg_word_size = sizeof(u16), -- 1.7.6.3

2 1

[alsa-devel] [PATCH] ASoC: Refcount WM8996 bandgap from FLL too
by Mark Brown 20 Sep '11

20 Sep '11

For digital only paths we need to make sure the bandgap is enabled prior to starting the FLL which isn't tied into DAPM. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 46 +++++++++++++++++++++++++++++++++++++++----- 1 files changed, 40 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index c587b4d..071dda0 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -72,6 +72,7 @@ struct wm8996_priv { struct regulator_bulk_data supplies[WM8996_NUM_SUPPLIES]; struct notifier_block disable_nb[WM8996_NUM_SUPPLIES]; struct regulator *cpvdd; + int bg_ena; struct wm8996_pdata pdata; @@ -667,14 +668,40 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static void wm8996_bg_enable(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + + wm8996->bg_ena++; + if (wm8996->bg_ena == 1) { + snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, + WM8996_BG_ENA, WM8996_BG_ENA); + msleep(2); + } +} + +static void wm8996_bg_disable(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + + wm8996->bg_ena--; + if (!wm8996->bg_ena) + snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, + WM8996_BG_ENA, 0); +} + static int bg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct snd_soc_codec *codec = w->codec; int ret = 0; switch (event) { - case SND_SOC_DAPM_POST_PMU: - msleep(2); + case SND_SOC_DAPM_PRE_PMU: + wm8996_bg_enable(codec); + break; + case SND_SOC_DAPM_POST_PMD: + wm8996_bg_disable(codec); break; default: BUG(); @@ -1015,10 +1042,10 @@ SND_SOC_DAPM_SUPPLY_S("SYSCLK", 1, WM8996_AIF_CLOCKING_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("SYSDSPCLK", 2, WM8996_CLOCKING_1, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_SUPPLY("Bandgap", WM8996_POWER_MANAGEMENT_1, WM8996_BG_ENA_SHIFT, - 0, bg_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_SUPPLY("Bandgap", SND_SOC_NOPM, 0, 0, bg_event, + SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0), @@ -2095,6 +2122,8 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_update_bits(codec, WM8996_FLL_CONTROL_1, WM8996_FLL_ENA, 0); + wm8996_bg_disable(codec); + return 0; } @@ -2149,6 +2178,11 @@ static int wm8996_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8996_FLL_EFS_1, fll_div.lambda); + /* Enable the bandgap if it's not already enabled */ + ret = snd_soc_read(codec, WM8996_FLL_CONTROL_1); + if (!(ret & WM8996_FLL_ENA)) + wm8996_bg_enable(codec); + /* Clear any pending completions (eg, from failed startups) */ try_wait_for_completion(&wm8996->fll_lock); -- 1.7.6.3

2 1

[alsa-devel] [PATCH] alsa-lib: if card driver name is empty string, use card name instead
by Scott Jiang 20 Sep '11

20 Sep '11

Some drivers don't have a card driver name, but they have a nice card name containing enough info. So we can use card name to search card config if we found driver name is not valid. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- src/confmisc.c | 8 +++++++- 1 files changed, 7 insertions(+), 1 deletions(-) diff --git a/src/confmisc.c b/src/confmisc.c index 80b0027..3c4b5aa 100644 --- a/src/confmisc.c +++ b/src/confmisc.c @@ -666,6 +666,7 @@ int snd_determine_driver(int card, char **driver) snd_ctl_t *ctl = NULL; snd_ctl_card_info_t *info; char *res = NULL; + const char *name = NULL; int err; assert(card >= 0 && card <= 32); @@ -680,7 +681,12 @@ int snd_determine_driver(int card, char **driver) SNDERR("snd_ctl_card_info error: %s", snd_strerror(err)); goto __error; } - res = strdup(snd_ctl_card_info_get_driver(info)); + name = snd_ctl_card_info_get_driver(info); + /* if card driver name is empty string, try to use card name */ + if (name[0] == '\0') + name = snd_ctl_card_info_get_name(info); + + res = strdup(name); if (res == NULL) err = -ENOMEM; else { -- 1.7.0.4

3 12

[PATCH] ALSA: HDA: Refactor Realtek's automute
by David Henningsson 20 Sep '11

20 Sep '11

Increase readability and understandability in the automute code. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/alc262_quirks.c | 28 +++------ sound/pci/hda/alc880_quirks.c | 17 ++--- sound/pci/hda/alc882_quirks.c | 85 +++++++++----------------- sound/pci/hda/alc_quirks.c | 9 +++ sound/pci/hda/patch_realtek.c | 136 ++++++++++++++++++++++------------------- 5 files changed, 127 insertions(+), 148 deletions(-) diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index c37e0c2..64d3ceb 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -61,10 +61,6 @@ static const struct snd_kcontrol_new alc262_base_mixer[] = { }; /* bind hp and internal speaker mute (with plug check) as master switch */ -static void alc262_hippo_master_update(struct hda_codec *codec) -{ - update_speakers(codec); -} static int alc262_hippo_master_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -85,7 +81,7 @@ static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol, if (val == spec->master_mute) return 0; spec->master_mute = val; - alc262_hippo_master_update(codec); + update_outputs(codec); return 1; } @@ -147,8 +143,7 @@ static void alc262_hippo_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc262_hippo1_setup(struct hda_codec *codec) @@ -157,8 +152,7 @@ static void alc262_hippo1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } @@ -221,8 +215,7 @@ static void alc262_tyan_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } @@ -364,8 +357,7 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec) spec->ext_mic_pin = 0x18; spec->int_mic_pin = 0x12; spec->auto_mic = 1; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_PIN, 1, 0, 0, 1); } /* @@ -446,8 +438,7 @@ static void alc262_fujitsu_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.hp_pins[1] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } /* bind volumes of both NID 0x0c and 0x0d */ @@ -493,8 +484,7 @@ static void alc262_lenovo_3000_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { @@ -599,8 +589,8 @@ static void alc262_ultra_automute(struct hda_codec *codec) mute = 0; /* auto-mute only when HP is used as HP */ if (!spec->cur_mux[0]) { - spec->jack_present = snd_hda_jack_detect(codec, 0x15); - if (spec->jack_present) + spec->hp_jack_present = snd_hda_jack_detect(codec, 0x15); + if (spec->hp_jack_present) mute = HDA_AMP_MUTE; } /* mute/unmute internal speaker */ diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index c844d2b..d051d856 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -749,8 +749,7 @@ static void alc880_uniwill_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc880_uniwill_init_hook(struct hda_codec *codec) @@ -781,8 +780,7 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) @@ -1051,8 +1049,7 @@ static void alc880_lg_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } /* @@ -1137,8 +1134,7 @@ static void alc880_lg_lw_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = { @@ -1188,7 +1184,7 @@ static void alc880_medion_rim_automute(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_hp_automute(codec); /* toggle EAPD */ - if (spec->jack_present) + if (spec->hp_jack_present) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); else snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2); @@ -1210,8 +1206,7 @@ static void alc880_medion_rim_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } #ifdef CONFIG_SND_HDA_POWER_SAVE diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 617d047..3252fd0 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -173,8 +173,7 @@ static void alc889_automute_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x17; spec->autocfg.speaker_pins[3] = 0x19; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc889_intel_init_hook(struct hda_codec *codec) @@ -191,8 +190,7 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1b; /* hp */ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ spec->autocfg.speaker_pins[1] = 0x15; /* bass */ - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } /* @@ -475,8 +473,7 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) @@ -487,8 +484,7 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) @@ -499,8 +495,7 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) @@ -511,8 +506,7 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } #define ALC882_DIGOUT_NID 0x06 @@ -1711,8 +1705,7 @@ static void alc885_imac24_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } #define alc885_mb5_setup alc885_imac24_setup @@ -1721,12 +1714,11 @@ static void alc885_imac24_setup(struct hda_codec *codec) /* Macbook Air 2,1 */ static void alc885_mba21_setup(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; + struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x18; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } @@ -1737,8 +1729,7 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc885_imac91_setup(struct hda_codec *codec) @@ -1748,8 +1739,7 @@ static void alc885_imac91_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static const struct hda_verb alc882_targa_verbs[] = { @@ -1773,7 +1763,7 @@ static void alc882_targa_automute(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_hp_automute(codec); snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, - spec->jack_present ? 1 : 3); + spec->hp_jack_present ? 1 : 3); } static void alc882_targa_setup(struct hda_codec *codec) @@ -1782,8 +1772,7 @@ static void alc882_targa_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) @@ -2187,8 +2176,7 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1a; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { @@ -2341,8 +2329,7 @@ static void alc883_mitac_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static const struct hda_verb alc883_mitac_verbs[] = { @@ -2507,8 +2494,7 @@ static void alc888_3st_hp_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x18; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static const struct hda_verb alc888_3st_hp_verbs[] = { @@ -2568,8 +2554,7 @@ static void alc888_lenovo_ms7195_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } /* toggle speaker-output according to the hp-jack state */ @@ -2579,8 +2564,7 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } /* toggle speaker-output according to the hp-jack state */ @@ -2593,8 +2577,7 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc883_clevo_m720_init_hook(struct hda_codec *codec) @@ -2623,8 +2606,7 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc883_haier_w66_setup(struct hda_codec *codec) @@ -2633,8 +2615,7 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc883_lenovo_101e_setup(struct hda_codec *codec) @@ -2644,10 +2625,7 @@ static void alc883_lenovo_101e_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->detect_line = 1; - spec->automute_lines = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 1, 1, 1); } /* toggle speaker-output according to the hp-jack state */ @@ -2658,8 +2636,7 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static const struct hda_verb alc883_acer_eapd_verbs[] = { @@ -2689,8 +2666,7 @@ static void alc888_6st_dell_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x15; spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc888_lenovo_sky_setup(struct hda_codec *codec) @@ -2703,8 +2679,7 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static void alc883_vaiott_setup(struct hda_codec *codec) @@ -2714,8 +2689,7 @@ static void alc883_vaiott_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static const struct hda_verb alc888_asus_m90v_verbs[] = { @@ -2739,8 +2713,7 @@ static void alc883_mode2_setup(struct hda_codec *codec) spec->ext_mic_pin = 0x18; spec->int_mic_pin = 0x19; spec->auto_mic = 1; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP, 1, 0, 0, 1); } static const struct hda_verb alc888_asus_eee1601_verbs[] = { diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c index 2be1129..e353333 100644 --- a/sound/pci/hda/alc_quirks.c +++ b/sound/pci/hda/alc_quirks.c @@ -453,6 +453,15 @@ static void setup_preset(struct hda_codec *codec, alc_fixup_autocfg_pin_nums(codec); } +static inline void alc_simple_setup_automute(struct alc_spec *spec, int mode, + int from_hp, int from_lo, int to_lo, int to_speaker) +{ + spec->automute_mode = mode; + spec->detect_hp = from_hp; + spec->detect_lo = from_lo; + spec->automute_lo = spec->automute_lo_possible = to_lo; + spec->automute_speaker = spec->automute_speaker_possible = to_speaker; +} /* auto-toggle front mic */ static void alc88x_simple_mic_automute(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b3c89c..de9a26b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -162,15 +162,17 @@ struct alc_spec { void (*automute_hook)(struct hda_codec *codec); /* for pin sensing */ - unsigned int jack_present: 1; + unsigned int hp_jack_present:1; unsigned int line_jack_present:1; unsigned int master_mute:1; unsigned int auto_mic:1; unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */ - unsigned int automute:1; /* HP automute enabled */ - unsigned int detect_line:1; /* Line-out detection enabled */ - unsigned int automute_lines:1; /* automute line-out as well; NOP when automute_hp_lo isn't set */ - unsigned int automute_hp_lo:1; /* both HP and LO available */ + unsigned int automute_speaker:1; /* automute speaker outputs */ + unsigned int automute_lo:1; /* automute LO outputs */ + unsigned int detect_hp:1; /* Headphone detection enabled */ + unsigned int detect_lo:1; /* Line-out detection enabled */ + unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */ + unsigned int automute_lo_possible:1; /* there are line outs and HP */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -530,8 +532,8 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } } -/* Toggle internal speakers muting */ -static void update_speakers(struct hda_codec *codec) +/* Toggle outputs muting */ +static void update_outputs(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int on; @@ -543,10 +545,10 @@ static void update_speakers(struct hda_codec *codec) do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins, spec->master_mute, true); - if (!spec->automute) + if (!spec->automute_speaker) on = 0; else - on = spec->jack_present | spec->line_jack_present; + on = spec->hp_jack_present | spec->line_jack_present; on |= spec->master_mute; do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins), spec->autocfg.speaker_pins, on, false); @@ -556,22 +558,22 @@ static void update_speakers(struct hda_codec *codec) if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] || spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0]) return; - if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines)) + if (!spec->automute_lo) on = 0; else - on = spec->jack_present; + on = spec->hp_jack_present; on |= spec->master_mute; do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins, on, false); } -static void call_update_speakers(struct hda_codec *codec) +static void call_update_outputs(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; if (spec->automute_hook) spec->automute_hook(codec); else - update_speakers(codec); + update_outputs(codec); } /* standard HP-automute helper */ @@ -579,12 +581,12 @@ static void alc_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->jack_present = + spec->hp_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); - if (!spec->automute) + if (!spec->detect_hp || (!spec->automute_speaker && !spec->automute_lo)) return; - call_update_speakers(codec); + call_update_outputs(codec); } /* standard line-out-automute helper */ @@ -595,9 +597,9 @@ static void alc_line_automute(struct hda_codec *codec) spec->line_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins); - if (!spec->automute || !spec->detect_line) + if (!spec->automute_speaker || !spec->detect_lo) return; - call_update_speakers(codec); + call_update_outputs(codec); } #define get_connection_index(codec, mux, nid) \ @@ -795,7 +797,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - if (spec->automute_hp_lo) { + if (spec->automute_speaker_possible && spec->automute_lo_possible) { uinfo->value.enumerated.items = 3; texts = texts3; } else { @@ -814,13 +816,12 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - unsigned int val; - if (!spec->automute) - val = 0; - else if (!spec->automute_hp_lo || !spec->automute_lines) - val = 1; - else - val = 2; + unsigned int val = 0; + if (spec->automute_speaker) + val++; + if (spec->automute_lo) + val++; + ucontrol->value.enumerated.item[0] = val; return 0; } @@ -833,29 +834,36 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, switch (ucontrol->value.enumerated.item[0]) { case 0: - if (!spec->automute) + if (!spec->automute_speaker && !spec->automute_lo) return 0; - spec->automute = 0; + spec->automute_speaker = 0; + spec->automute_lo = 0; break; case 1: - if (spec->automute && - (!spec->automute_hp_lo || !spec->automute_lines)) - return 0; - spec->automute = 1; - spec->automute_lines = 0; + if (spec->automute_speaker_possible) { + if (!spec->automute_lo && spec->automute_speaker) + return 0; + spec->automute_speaker = 1; + spec->automute_lo = 0; + } else if (spec->automute_lo_possible) { + if (spec->automute_lo) + return 0; + spec->automute_lo = 1; + } else + return -EINVAL; break; case 2: - if (!spec->automute_hp_lo) + if (!spec->automute_lo_possible || !spec->automute_speaker_possible) return -EINVAL; - if (spec->automute && spec->automute_lines) + if (spec->automute_speaker && spec->automute_lo) return 0; - spec->automute = 1; - spec->automute_lines = 1; + spec->automute_speaker = 1; + spec->automute_lo = 1; break; default: return -EINVAL; } - call_update_speakers(codec); + call_update_outputs(codec); return 1; } @@ -892,7 +900,7 @@ static int alc_add_automute_mode_enum(struct hda_codec *codec) * Check the availability of HP/line-out auto-mute; * Set up appropriately if really supported */ -static void alc_init_auto_hp(struct hda_codec *codec) +static void alc_init_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; @@ -907,8 +915,6 @@ static void alc_init_auto_hp(struct hda_codec *codec) present++; if (present < 2) /* need two different output types */ return; - if (present == 3) - spec->automute_hp_lo = 1; /* both HP and LO automute */ if (!cfg->speaker_pins[0] && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { @@ -924,6 +930,8 @@ static void alc_init_auto_hp(struct hda_codec *codec) cfg->hp_outs = cfg->line_outs; } + spec->automute_mode = ALC_AUTOMUTE_PIN; + for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; if (!is_jack_detectable(codec, nid)) @@ -933,28 +941,32 @@ static void alc_init_auto_hp(struct hda_codec *codec) snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT); - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; - } - if (spec->automute && cfg->line_out_pins[0] && - cfg->speaker_pins[0] && - cfg->line_out_pins[0] != cfg->hp_pins[0] && - cfg->line_out_pins[0] != cfg->speaker_pins[0]) { - for (i = 0; i < cfg->line_outs; i++) { - hda_nid_t nid = cfg->line_out_pins[i]; - if (!is_jack_detectable(codec, nid)) - continue; - snd_printdd("realtek: Enable Line-Out auto-muting " - "on NID 0x%x\n", nid); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC_FRONT_EVENT); - spec->detect_line = 1; + spec->detect_hp = 1; + } + + if (cfg->line_out_type == AUTO_PIN_LINE_OUT && cfg->line_outs) { + if (cfg->speaker_outs) + for (i = 0; i < cfg->line_outs; i++) { + hda_nid_t nid = cfg->line_out_pins[i]; + if (!is_jack_detectable(codec, nid)) + continue; + snd_printdd("realtek: Enable Line-Out " + "auto-muting on NID 0x%x\n", nid); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC_FRONT_EVENT); + spec->detect_lo = 1; } - spec->automute_lines = spec->detect_line; + spec->automute_lo_possible = spec->detect_hp; } - if (spec->automute) { + spec->automute_speaker_possible = cfg->speaker_outs && + (spec->detect_hp || spec->detect_lo); + + spec->automute_lo = spec->automute_lo_possible; + spec->automute_speaker = spec->automute_speaker_possible; + + if (spec->automute_speaker_possible || spec->automute_lo_possible) { /* create a control for automute mode */ alc_add_automute_mode_enum(codec); spec->unsol_event = alc_sku_unsol_event; @@ -1155,7 +1167,7 @@ static void alc_init_auto_mic(struct hda_codec *codec) /* check the availabilities of auto-mute and auto-mic switches */ static void alc_auto_check_switches(struct hda_codec *codec) { - alc_init_auto_hp(codec); + alc_init_automute(codec); alc_init_auto_mic(codec); } @@ -4641,7 +4653,7 @@ static void alc269_fixup_stereo_dmic(struct hda_codec *codec, static void alc269_quanta_automute(struct hda_codec *codec) { - update_speakers(codec); + update_outputs(codec); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); -- 1.7.5.4 --------------000109060900060009060200--

1 0

[PATCH] ALSA: HDA: Refactor Realtek's automute
by David Henningsson 20 Sep '11

20 Sep '11

Increase readability and understandability in the automute code. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/alc262_quirks.c | 28 +++------ sound/pci/hda/alc880_quirks.c | 17 ++--- sound/pci/hda/alc882_quirks.c | 85 +++++++++----------------- sound/pci/hda/alc_quirks.c | 13 ++++ sound/pci/hda/patch_realtek.c | 136 ++++++++++++++++++++++------------------- 5 files changed, 131 insertions(+), 148 deletions(-) diff --git a/sound/pci/hda/alc262_quirks.c b/sound/pci/hda/alc262_quirks.c index c37e0c2..7894b2b 100644 --- a/sound/pci/hda/alc262_quirks.c +++ b/sound/pci/hda/alc262_quirks.c @@ -61,10 +61,6 @@ static const struct snd_kcontrol_new alc262_base_mixer[] = { }; /* bind hp and internal speaker mute (with plug check) as master switch */ -static void alc262_hippo_master_update(struct hda_codec *codec) -{ - update_speakers(codec); -} static int alc262_hippo_master_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -85,7 +81,7 @@ static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol, if (val == spec->master_mute) return 0; spec->master_mute = val; - alc262_hippo_master_update(codec); + update_outputs(codec); return 1; } @@ -147,8 +143,7 @@ static void alc262_hippo_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc262_hippo1_setup(struct hda_codec *codec) @@ -157,8 +152,7 @@ static void alc262_hippo1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } @@ -221,8 +215,7 @@ static void alc262_tyan_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } @@ -364,8 +357,7 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec) spec->ext_mic_pin = 0x18; spec->int_mic_pin = 0x12; spec->auto_mic = 1; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_PIN); } /* @@ -446,8 +438,7 @@ static void alc262_fujitsu_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.hp_pins[1] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* bind volumes of both NID 0x0c and 0x0d */ @@ -493,8 +484,7 @@ static void alc262_lenovo_3000_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { @@ -599,8 +589,8 @@ static void alc262_ultra_automute(struct hda_codec *codec) mute = 0; /* auto-mute only when HP is used as HP */ if (!spec->cur_mux[0]) { - spec->jack_present = snd_hda_jack_detect(codec, 0x15); - if (spec->jack_present) + spec->hp_jack_present = snd_hda_jack_detect(codec, 0x15); + if (spec->hp_jack_present) mute = HDA_AMP_MUTE; } /* mute/unmute internal speaker */ diff --git a/sound/pci/hda/alc880_quirks.c b/sound/pci/hda/alc880_quirks.c index c844d2b..bea22ed 100644 --- a/sound/pci/hda/alc880_quirks.c +++ b/sound/pci/hda/alc880_quirks.c @@ -749,8 +749,7 @@ static void alc880_uniwill_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc880_uniwill_init_hook(struct hda_codec *codec) @@ -781,8 +780,7 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) @@ -1051,8 +1049,7 @@ static void alc880_lg_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* @@ -1137,8 +1134,7 @@ static void alc880_lg_lw_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct snd_kcontrol_new alc880_medion_rim_mixer[] = { @@ -1188,7 +1184,7 @@ static void alc880_medion_rim_automute(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_hp_automute(codec); /* toggle EAPD */ - if (spec->jack_present) + if (spec->hp_jack_present) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 0); else snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, 2); @@ -1210,8 +1206,7 @@ static void alc880_medion_rim_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } #ifdef CONFIG_SND_HDA_POWER_SAVE diff --git a/sound/pci/hda/alc882_quirks.c b/sound/pci/hda/alc882_quirks.c index 617d047..e251514 100644 --- a/sound/pci/hda/alc882_quirks.c +++ b/sound/pci/hda/alc882_quirks.c @@ -173,8 +173,7 @@ static void alc889_automute_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x17; spec->autocfg.speaker_pins[3] = 0x19; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc889_intel_init_hook(struct hda_codec *codec) @@ -191,8 +190,7 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1b; /* hp */ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ spec->autocfg.speaker_pins[1] = 0x15; /* bass */ - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* @@ -475,8 +473,7 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) @@ -487,8 +484,7 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) @@ -499,8 +495,7 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) @@ -511,8 +506,7 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } #define ALC882_DIGOUT_NID 0x06 @@ -1711,8 +1705,7 @@ static void alc885_imac24_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } #define alc885_mb5_setup alc885_imac24_setup @@ -1721,12 +1714,11 @@ static void alc885_imac24_setup(struct hda_codec *codec) /* Macbook Air 2,1 */ static void alc885_mba21_setup(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; + struct alc_spec *spec = codec->spec; - spec->autocfg.hp_pins[0] = 0x14; - spec->autocfg.speaker_pins[0] = 0x18; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + spec->autocfg.hp_pins[0] = 0x14; + spec->autocfg.speaker_pins[0] = 0x18; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } @@ -1737,8 +1729,7 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc885_imac91_setup(struct hda_codec *codec) @@ -1748,8 +1739,7 @@ static void alc885_imac91_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc882_targa_verbs[] = { @@ -1773,7 +1763,7 @@ static void alc882_targa_automute(struct hda_codec *codec) struct alc_spec *spec = codec->spec; alc_hp_automute(codec); snd_hda_codec_write_cache(codec, 1, 0, AC_VERB_SET_GPIO_DATA, - spec->jack_present ? 1 : 3); + spec->hp_jack_present ? 1 : 3); } static void alc882_targa_setup(struct hda_codec *codec) @@ -1782,8 +1772,7 @@ static void alc882_targa_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res) @@ -2187,8 +2176,7 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1a; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { @@ -2341,8 +2329,7 @@ static void alc883_mitac_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc883_mitac_verbs[] = { @@ -2507,8 +2494,7 @@ static void alc888_3st_hp_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x18; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc888_3st_hp_verbs[] = { @@ -2568,8 +2554,7 @@ static void alc888_lenovo_ms7195_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* toggle speaker-output according to the hp-jack state */ @@ -2579,8 +2564,7 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* toggle speaker-output according to the hp-jack state */ @@ -2593,8 +2577,7 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc883_clevo_m720_init_hook(struct hda_codec *codec) @@ -2623,8 +2606,7 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc883_haier_w66_setup(struct hda_codec *codec) @@ -2633,8 +2615,7 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc883_lenovo_101e_setup(struct hda_codec *codec) @@ -2644,10 +2625,7 @@ static void alc883_lenovo_101e_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; - spec->detect_line = 1; - spec->automute_lines = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } /* toggle speaker-output according to the hp-jack state */ @@ -2658,8 +2636,7 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc883_acer_eapd_verbs[] = { @@ -2689,8 +2666,7 @@ static void alc888_6st_dell_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x15; spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc888_lenovo_sky_setup(struct hda_codec *codec) @@ -2703,8 +2679,7 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static void alc883_vaiott_setup(struct hda_codec *codec) @@ -2714,8 +2689,7 @@ static void alc883_vaiott_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc888_asus_m90v_verbs[] = { @@ -2739,8 +2713,7 @@ static void alc883_mode2_setup(struct hda_codec *codec) spec->ext_mic_pin = 0x18; spec->int_mic_pin = 0x19; spec->auto_mic = 1; - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_AMP; + alc_simple_setup_automute(spec, ALC_AUTOMUTE_AMP); } static const struct hda_verb alc888_asus_eee1601_verbs[] = { diff --git a/sound/pci/hda/alc_quirks.c b/sound/pci/hda/alc_quirks.c index 2be1129..a18952e 100644 --- a/sound/pci/hda/alc_quirks.c +++ b/sound/pci/hda/alc_quirks.c @@ -453,6 +453,19 @@ static void setup_preset(struct hda_codec *codec, alc_fixup_autocfg_pin_nums(codec); } +static void alc_simple_setup_automute(struct alc_spec *spec, int mode) +{ + int lo_pin = spec->autocfg.line_out_pins[0]; + + if (lo_pin == spec->autocfg.speaker_pins[0] || + lo_pin == spec->autocfg.hp_pins[0]) + lo_pin = 0; + spec->automute_mode = mode; + spec->detect_hp = !!spec->autocfg.hp_pins[0]; + spec->detect_lo = !!lo_pin; + spec->automute_lo = spec->automute_lo_possible = !!lo_pin; + spec->automute_speaker = spec->automute_speaker_possible = !!spec->autocfg.speaker_pins[0]; +} /* auto-toggle front mic */ static void alc88x_simple_mic_automute(struct hda_codec *codec) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b3c89c..de9a26b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -162,15 +162,17 @@ struct alc_spec { void (*automute_hook)(struct hda_codec *codec); /* for pin sensing */ - unsigned int jack_present: 1; + unsigned int hp_jack_present:1; unsigned int line_jack_present:1; unsigned int master_mute:1; unsigned int auto_mic:1; unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */ - unsigned int automute:1; /* HP automute enabled */ - unsigned int detect_line:1; /* Line-out detection enabled */ - unsigned int automute_lines:1; /* automute line-out as well; NOP when automute_hp_lo isn't set */ - unsigned int automute_hp_lo:1; /* both HP and LO available */ + unsigned int automute_speaker:1; /* automute speaker outputs */ + unsigned int automute_lo:1; /* automute LO outputs */ + unsigned int detect_hp:1; /* Headphone detection enabled */ + unsigned int detect_lo:1; /* Line-out detection enabled */ + unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */ + unsigned int automute_lo_possible:1; /* there are line outs and HP */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -530,8 +532,8 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins, } } -/* Toggle internal speakers muting */ -static void update_speakers(struct hda_codec *codec) +/* Toggle outputs muting */ +static void update_outputs(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int on; @@ -543,10 +545,10 @@ static void update_speakers(struct hda_codec *codec) do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins, spec->master_mute, true); - if (!spec->automute) + if (!spec->automute_speaker) on = 0; else - on = spec->jack_present | spec->line_jack_present; + on = spec->hp_jack_present | spec->line_jack_present; on |= spec->master_mute; do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins), spec->autocfg.speaker_pins, on, false); @@ -556,22 +558,22 @@ static void update_speakers(struct hda_codec *codec) if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] || spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0]) return; - if (!spec->automute || (spec->automute_hp_lo && !spec->automute_lines)) + if (!spec->automute_lo) on = 0; else - on = spec->jack_present; + on = spec->hp_jack_present; on |= spec->master_mute; do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins, on, false); } -static void call_update_speakers(struct hda_codec *codec) +static void call_update_outputs(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; if (spec->automute_hook) spec->automute_hook(codec); else - update_speakers(codec); + update_outputs(codec); } /* standard HP-automute helper */ @@ -579,12 +581,12 @@ static void alc_hp_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - spec->jack_present = + spec->hp_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); - if (!spec->automute) + if (!spec->detect_hp || (!spec->automute_speaker && !spec->automute_lo)) return; - call_update_speakers(codec); + call_update_outputs(codec); } /* standard line-out-automute helper */ @@ -595,9 +597,9 @@ static void alc_line_automute(struct hda_codec *codec) spec->line_jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.line_out_pins), spec->autocfg.line_out_pins); - if (!spec->automute || !spec->detect_line) + if (!spec->automute_speaker || !spec->detect_lo) return; - call_update_speakers(codec); + call_update_outputs(codec); } #define get_connection_index(codec, mux, nid) \ @@ -795,7 +797,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - if (spec->automute_hp_lo) { + if (spec->automute_speaker_possible && spec->automute_lo_possible) { uinfo->value.enumerated.items = 3; texts = texts3; } else { @@ -814,13 +816,12 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - unsigned int val; - if (!spec->automute) - val = 0; - else if (!spec->automute_hp_lo || !spec->automute_lines) - val = 1; - else - val = 2; + unsigned int val = 0; + if (spec->automute_speaker) + val++; + if (spec->automute_lo) + val++; + ucontrol->value.enumerated.item[0] = val; return 0; } @@ -833,29 +834,36 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, switch (ucontrol->value.enumerated.item[0]) { case 0: - if (!spec->automute) + if (!spec->automute_speaker && !spec->automute_lo) return 0; - spec->automute = 0; + spec->automute_speaker = 0; + spec->automute_lo = 0; break; case 1: - if (spec->automute && - (!spec->automute_hp_lo || !spec->automute_lines)) - return 0; - spec->automute = 1; - spec->automute_lines = 0; + if (spec->automute_speaker_possible) { + if (!spec->automute_lo && spec->automute_speaker) + return 0; + spec->automute_speaker = 1; + spec->automute_lo = 0; + } else if (spec->automute_lo_possible) { + if (spec->automute_lo) + return 0; + spec->automute_lo = 1; + } else + return -EINVAL; break; case 2: - if (!spec->automute_hp_lo) + if (!spec->automute_lo_possible || !spec->automute_speaker_possible) return -EINVAL; - if (spec->automute && spec->automute_lines) + if (spec->automute_speaker && spec->automute_lo) return 0; - spec->automute = 1; - spec->automute_lines = 1; + spec->automute_speaker = 1; + spec->automute_lo = 1; break; default: return -EINVAL; } - call_update_speakers(codec); + call_update_outputs(codec); return 1; } @@ -892,7 +900,7 @@ static int alc_add_automute_mode_enum(struct hda_codec *codec) * Check the availability of HP/line-out auto-mute; * Set up appropriately if really supported */ -static void alc_init_auto_hp(struct hda_codec *codec) +static void alc_init_automute(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; @@ -907,8 +915,6 @@ static void alc_init_auto_hp(struct hda_codec *codec) present++; if (present < 2) /* need two different output types */ return; - if (present == 3) - spec->automute_hp_lo = 1; /* both HP and LO automute */ if (!cfg->speaker_pins[0] && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { @@ -924,6 +930,8 @@ static void alc_init_auto_hp(struct hda_codec *codec) cfg->hp_outs = cfg->line_outs; } + spec->automute_mode = ALC_AUTOMUTE_PIN; + for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; if (!is_jack_detectable(codec, nid)) @@ -933,28 +941,32 @@ static void alc_init_auto_hp(struct hda_codec *codec) snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT); - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; - } - if (spec->automute && cfg->line_out_pins[0] && - cfg->speaker_pins[0] && - cfg->line_out_pins[0] != cfg->hp_pins[0] && - cfg->line_out_pins[0] != cfg->speaker_pins[0]) { - for (i = 0; i < cfg->line_outs; i++) { - hda_nid_t nid = cfg->line_out_pins[i]; - if (!is_jack_detectable(codec, nid)) - continue; - snd_printdd("realtek: Enable Line-Out auto-muting " - "on NID 0x%x\n", nid); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC_FRONT_EVENT); - spec->detect_line = 1; + spec->detect_hp = 1; + } + + if (cfg->line_out_type == AUTO_PIN_LINE_OUT && cfg->line_outs) { + if (cfg->speaker_outs) + for (i = 0; i < cfg->line_outs; i++) { + hda_nid_t nid = cfg->line_out_pins[i]; + if (!is_jack_detectable(codec, nid)) + continue; + snd_printdd("realtek: Enable Line-Out " + "auto-muting on NID 0x%x\n", nid); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC_FRONT_EVENT); + spec->detect_lo = 1; } - spec->automute_lines = spec->detect_line; + spec->automute_lo_possible = spec->detect_hp; } - if (spec->automute) { + spec->automute_speaker_possible = cfg->speaker_outs && + (spec->detect_hp || spec->detect_lo); + + spec->automute_lo = spec->automute_lo_possible; + spec->automute_speaker = spec->automute_speaker_possible; + + if (spec->automute_speaker_possible || spec->automute_lo_possible) { /* create a control for automute mode */ alc_add_automute_mode_enum(codec); spec->unsol_event = alc_sku_unsol_event; @@ -1155,7 +1167,7 @@ static void alc_init_auto_mic(struct hda_codec *codec) /* check the availabilities of auto-mute and auto-mic switches */ static void alc_auto_check_switches(struct hda_codec *codec) { - alc_init_auto_hp(codec); + alc_init_automute(codec); alc_init_auto_mic(codec); } @@ -4641,7 +4653,7 @@ static void alc269_fixup_stereo_dmic(struct hda_codec *codec, static void alc269_quanta_automute(struct hda_codec *codec) { - update_speakers(codec); + update_outputs(codec); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); -- 1.7.5.4 --------------040802070305030607060106--

1 0

[alsa-devel] [PATCH] pulse: Install a PulseAudio config snippet into alsa.conf.d
by gmane＠colin.guthr.ie 20 Sep '11

20 Sep '11

From: Colin Guthrie <colin(a)mageia.org> v2: Assing priorities to the config snippets. Signed-off-by: Colin Guthrie <colin(a)mageia.org> --- configure.in | 19 ++++++++++++++++++- pulse/50-pulseaudio.conf | 13 +++++++++++++ pulse/99-pulseaudio-default.conf.example | 13 +++++++++++++ pulse/Makefile.am | 2 ++ 4 files changed, 46 insertions(+), 1 deletions(-) create mode 100644 pulse/50-pulseaudio.conf create mode 100644 pulse/99-pulseaudio-default.conf.example diff --git a/configure.in b/configure.in index ccf59ba..3839dd5 100644 --- a/configure.in +++ b/configure.in @@ -131,10 +131,10 @@ fi AM_CONDITIONAL(HAVE_PPH, test "$PPH" = "builtin" -o "$PPH" = "lib") AM_CONDITIONAL(USE_LIBSPEEX, test "$PPH" = "lib") -dnl ALSA plugin directory test "x$prefix" = xNONE && prefix=$ac_default_prefix test "x$exec_prefix" = xNONE && exec_prefix=$prefix +dnl ALSA plugin directory AC_ARG_WITH(plugindir, AS_HELP_STRING([--with-plugindir=dir], [path where ALSA plugin files are stored]), @@ -151,6 +151,23 @@ AC_DEFINE_UNQUOTED(ALSA_PLUGIN_DIR, "$plugindir", [directory containing ALSA add ALSA_PLUGIN_DIR="$plugindir" AC_SUBST(ALSA_PLUGIN_DIR) +dnl ALSA data directory +AC_ARG_WITH(alsadatadir, + AS_HELP_STRING([--with-alsadatadir=dir], + [path where ALSA data files are stored]), + alsadatadir="$withval", alsadatadir="") +if test -z "$alsadatadir"; then + eval dir="$datadir" + case "$dir" in + /*) ;; + *) dir="$dir" + esac + alsadatadir="$dir/alsa" +fi +AC_DEFINE_UNQUOTED(ALSA_DATA_DIR, "$alsadatadir", [directory containing ALSA data files]) +ALSA_DATA_DIR="$alsadatadir" +AC_SUBST(ALSA_DATA_DIR) + SAVE_PLUGINS_VERSION AC_OUTPUT([ diff --git a/pulse/50-pulseaudio.conf b/pulse/50-pulseaudio.conf new file mode 100644 index 0000000..dd85dab --- /dev/null +++ b/pulse/50-pulseaudio.conf @@ -0,0 +1,13 @@ +# Add a specific named PulseAudio pcm and ctl (typically useful for testing) + +pcm.pulse { + type pulse + hint { + show on + description "PulseAudio Sound Server" + } +} + +ctl.pulse { + type pulse +} diff --git a/pulse/99-pulseaudio-default.conf.example b/pulse/99-pulseaudio-default.conf.example new file mode 100644 index 0000000..4f58858 --- /dev/null +++ b/pulse/99-pulseaudio-default.conf.example @@ -0,0 +1,13 @@ +# Default to PulseAudio + +pcm.!default { + type pulse + hint { + show on + description "Default ALSA Output (currently PulseAudio Sound Server)" + } +} + +ctl.!default { + type pulse +} diff --git a/pulse/Makefile.am b/pulse/Makefile.am index e89e8ee..e184c62 100644 --- a/pulse/Makefile.am +++ b/pulse/Makefile.am @@ -1,10 +1,12 @@ asound_module_pcm_LTLIBRARIES = libasound_module_pcm_pulse.la asound_module_ctl_LTLIBRARIES = libasound_module_ctl_pulse.la asound_module_conf_LTLIBRARIES = libasound_module_conf_pulse.la +asound_module_data_DATA = 50-pulseaudio.conf 99-pulseaudio-default.conf.example asound_module_pcmdir = @ALSA_PLUGIN_DIR@ asound_module_ctldir = @ALSA_PLUGIN_DIR@ asound_module_confdir = @ALSA_PLUGIN_DIR@ +asound_module_datadir = @ALSA_DATA_DIR@/alsa.conf.d AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ $(PTHREAD_CFLAGS) $(pulseaudio_CFLAGS) -D_GNU_SOURCE AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED) -- 1.7.6

3 2

[alsa-devel] [PATCH] conf: Allow for a directory to be given as a config file.
by gmane＠colin.guthr.ie 20 Sep '11

20 Sep '11

From: Colin Guthrie <colin(a)mageia.org> When this is done, *.conf files can be placed in that directory and they will be processed by as if they were included directly. A directory (typically /usr/share/alsa/alsa.conf.d/) has been added into the distribution. v2: Used existing conf syntax rather than processing via autotools v3: Split file loading into separate function and made error handling more consistent. Signed-off-by: Colin Guthrie <colin(a)mageia.org> --- configure.in | 3 +- src/conf.c | 78 ++++++++++++++++++++++++++++++++----- src/conf/Makefile.am | 2 +- src/conf/alsa.conf | 7 +++ src/conf/alsa.conf.d/Makefile.am | 8 ++++ src/conf/alsa.conf.d/README | 2 + 6 files changed, 87 insertions(+), 13 deletions(-) create mode 100644 src/conf/alsa.conf.d/Makefile.am create mode 100644 src/conf/alsa.conf.d/README diff --git a/configure.in b/configure.in index 7ee0ccc..13e38b8 100644 --- a/configure.in +++ b/configure.in @@ -616,7 +616,8 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/pcm/Makefile src/pcm/scopes/Makefile \ src/rawmidi/Makefile src/timer/Makefile \ src/hwdep/Makefile src/seq/Makefile src/ucm/Makefile \ - src/compat/Makefile src/alisp/Makefile src/conf/Makefile \ + src/compat/Makefile src/alisp/Makefile \ + src/conf/Makefile src/conf/alsa.conf.d/Makefile \ src/conf/cards/Makefile \ src/conf/pcm/Makefile \ modules/Makefile modules/mixer/Makefile modules/mixer/simple/Makefile \ diff --git a/src/conf.c b/src/conf.c index ddefff6..5b1b5a6 100644 --- a/src/conf.c +++ b/src/conf.c @@ -417,6 +417,7 @@ beginning:</P> #include <stdarg.h> #include <limits.h> #include <sys/stat.h> +#include <dirent.h> #include <locale.h> #include "local.h" #ifdef HAVE_LIBPTHREAD @@ -3373,6 +3374,42 @@ static int snd_config_hooks(snd_config_t *config, snd_config_t *private_data) return err; } +static int config_filename_filter(const struct dirent *dirent) +{ + size_t flen; + + if (dirent == NULL) + return 0; + if (dirent->d_type == DT_DIR) + return 0; + + flen = strlen(dirent->d_name); + if (flen <= 5) + return 0; + + if (strncmp(&dirent->d_name[flen-5], ".conf", 5) == 0) + return 1; + + return 0; +} + +static int config_file_open(snd_config_t *root, const char *filename) +{ + snd_input_t *in; + int err; + + err = snd_input_stdio_open(&in, filename, "r"); + if (err >= 0) { + err = snd_config_load(root, in); + snd_input_close(in); + if (err < 0) + SNDERR("%s may be old or corrupted: consider to remove or fix it", filename); + } else + SNDERR("cannot access file %s", filename); + + return err; +} + /** * \brief Loads and parses the given configurations files. * \param[in] root Handle to the root configuration node. @@ -3457,20 +3494,39 @@ int snd_config_hook_load(snd_config_t *root, snd_config_t *config, snd_config_t } } while (hit); for (idx = 0; idx < fi_count; idx++) { - snd_input_t *in; + struct stat st; if (!errors && access(fi[idx].name, R_OK) < 0) continue; - err = snd_input_stdio_open(&in, fi[idx].name, "r"); - if (err >= 0) { - err = snd_config_load(root, in); - snd_input_close(in); - if (err < 0) { - SNDERR("%s may be old or corrupted: consider to remove or fix it", fi[idx].name); - goto _err; - } - } else { - SNDERR("cannot access file %s", fi[idx].name); + if (stat(fi[idx].name, &st) < 0) { + SNDERR("cannot stat file/directory %s", fi[idx].name); + continue; } + if (S_ISDIR(st.st_mode)) { + struct dirent **namelist; + int n; + + n = scandir(fi[idx].name, &namelist, config_filename_filter, versionsort); + if (n > 0) { + int j; + err = 0; + for (j = 0; j < n; ++j) { + if (err >= 0) { + int sl = strlen(fi[idx].name) + strlen(namelist[j]->d_name) + 2; + char *filename = malloc(sl); + snprintf(filename, sl, "%s/%s", fi[idx].name, namelist[j]->d_name); + filename[sl-1] = '\0'; + + err = config_file_open(root, filename); + free(filename); + } + free(namelist[j]); + } + free(namelist); + if (err < 0) + goto _err; + } + } else if (config_file_open(root, fi[idx].name) < 0) + goto _err; } *dst = NULL; err = 0; diff --git a/src/conf/Makefile.am b/src/conf/Makefile.am index 2e5d0bf..456454f 100644 --- a/src/conf/Makefile.am +++ b/src/conf/Makefile.am @@ -1,4 +1,4 @@ -SUBDIRS=cards pcm +SUBDIRS=cards pcm alsa.conf.d cfg_files = alsa.conf if BUILD_ALISP diff --git a/src/conf/alsa.conf b/src/conf/alsa.conf index a33c24e..bc91df3 100644 --- a/src/conf/alsa.conf +++ b/src/conf/alsa.conf @@ -8,6 +8,13 @@ { func load files [ + { + @func concat + strings [ + { @func datadir } + "/alsa.conf.d/" + ] + } "/etc/asound.conf" "~/.asoundrc" ] diff --git a/src/conf/alsa.conf.d/Makefile.am b/src/conf/alsa.conf.d/Makefile.am new file mode 100644 index 0000000..c91661e --- /dev/null +++ b/src/conf/alsa.conf.d/Makefile.am @@ -0,0 +1,8 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +alsadir = $(alsaconfigdir)/alsa.conf.d +cfg_files = README + +alsa_DATA = $(cfg_files) + +EXTRA_DIST = \ + $(cfg_files) diff --git a/src/conf/alsa.conf.d/README b/src/conf/alsa.conf.d/README new file mode 100644 index 0000000..9997884 --- /dev/null +++ b/src/conf/alsa.conf.d/README @@ -0,0 +1,2 @@ +You can place files named *.conf in this folder and they will be processed +when initialising alsa-lib. -- 1.7.6

2 1

[alsa-devel] [PATCH] ALSA: HDA: Add support for IDT 92HD93
by David Henningsson 20 Sep '11

20 Sep '11

Cc: stable(a)kernel.org BugLink: http://bugs.launchpad.net/bugs/854468 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 5145b66..7d8efab 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -6573,6 +6573,7 @@ static const struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx }, { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx }, { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx }, + { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx}, { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx}, -- 1.7.5.4

2 1

[alsa-devel] [PATCH 1/1] ymfpci - enable rear device and dup4ch control when ac97 support 4 channels
by Raymond Yau 20 Sep '11

20 Sep '11

ymfpci - enable rear device and dup4ch control when ac97 support 4 channels

2 3

Re: [alsa-devel] hda- Disable Sticky PCM
by Takashi Iwai 20 Sep '11

20 Sep '11

At Tue, 20 Sep 2011 08:15:09 +0800, Raymond Yau wrote: > > 2011/9/18 Takashi Iwai <tiwai(a)suse.de>: > > > > Well, I can't reproduce it in hda-emu, so I don't know now. > > Try to get a stacktrace to see what's calling there actually. > > > > > set stream active : active stream 1 > ALSA hda_intel.c:1757: azx_pcm_prepare: bufsize=0xd750, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x4, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x3, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x6, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x5, stream=0x5, > channel=0, format=0x4111 > ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0xa, stream=0x5, > channel=0, format=0x4111 > hda-intel: IRQ timing workaround is activated for card #1. Suggest a > bigger bdl_pos_adj. > ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x4 > ------------[ cut here ]------------ (snip) Thanks, so it's called from hw_free. This is a call just to be sure, and should be harmless. If you mind, the verbs are suppressed by a patch like below. Takashi --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 6b611d5..e3db196 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1491,8 +1491,11 @@ static void really_cleanup_stream(struct hda_codec *codec, struct hda_cvt_setup *q) { hda_nid_t nid = q->nid; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); + if (q->stream_tag || q->channel_id) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); + if (q->format_id) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0 +); memset(q, 0, sizeof(*q)); q->nid = nid; }

1 0

[alsa-devel] [PATCH] ASoC: Fix unused variable warning in WM8996
by Mark Brown 19 Sep '11

19 Sep '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 2 -- 1 files changed, 0 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f455997..bede1ac 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -670,8 +670,6 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, static int bg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = w->codec; - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); int ret = 0; switch (event) { -- 1.7.6.3

1 0

[PATCH 2/2] [ASoC] 88pm860x-codec - reset the codec correctly
by Bas Vermeulen 19 Sep '11

19 Sep '11

Reset the codec according to the Audio power-up delay errata for the 88PM8607. Signed-off-by: Bas Vermeulen <bas.vermeulen(a)novero.com> --- sound/soc/codecs/88pm860x-codec.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 0198dbb..df7b4a0 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1179,6 +1179,9 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_STANDBY: if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ + data = AUDIO_PLL | AUDIO_SECTION_ON; + pm860x_reg_write(codec->control_data, REG_MISC2, data); + udelay(300); data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON; pm860x_reg_write(codec->control_data, REG_MISC2, data); -- 1.7.0.4 --------------080107070107020807050104--

1 0

[PATCH 1/2] [ASoC] 88pm860x-codec - Allow independent use of both I2S playback and capture
by Bas Vermeulen 19 Sep '11

19 Sep '11

Introduce a I2S CLK supply so playback and capture can operate independently. Signed-off-by: Bas Vermeulen <bas.vermeulen(a)novero.com> --- sound/soc/codecs/88pm860x-codec.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 1924157..0198dbb 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -772,11 +772,12 @@ static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0, - PM860X_DAC_EN_2, 0, 0), + SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0, PM860X_I2S_IFACE_3, 5, 1), + SND_SOC_DAPM_SUPPLY("I2S CLK", PM860X_DAC_EN_2, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux), SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux), SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux), @@ -868,6 +869,11 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Left ADC", NULL, "Left ADC MOD"}, {"Right ADC", NULL, "Right ADC MOD"}, + /* I2S Clock */ + {"I2S DIN", NULL, "I2S CLK"}, + {"I2S DIN1", NULL, "I2S CLK"}, + {"I2S DOUT", NULL, "I2S CLK"}, + /* PCM/AIF1 Inputs */ {"PCM SDO", NULL, "ADC Left Mux"}, {"PCM SDO", NULL, "ADCR EC Mux"}, -- 1.7.0.4 --------------030808060208020409080204--

1 0

Re: [alsa-devel] [PATCH] ASoC: sn95031: Staticize sn95031_pcm_hw_params
by Mark Brown 19 Sep '11

19 Sep '11

On Mon, Sep 19, 2011 at 04:34:28PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks.

1 0

[alsa-devel] [RFC PATCH] Fixup automute for Realtek auto parser
by David Henningsson 19 Sep '11

19 Sep '11

The ~6 months old functionality for configurable automuting is broken for the case that the user has only HP and LO (no speakers) - basically there is an "Automute mode" control, but enabling it does nothing. While fixing that, I also took the liberty of refactoring/rewriting some of that code, in order to make it easier to understand and maintain. Takashi, given you approve/like these changes, I'll go ahead and fix up the quirk files to fit the new variable names. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 12

Re: [alsa-devel] [PATCH] ASoC: sn95031: Staticize sn95031_pcm_hw_params
by Vinod Koul 19 Sep '11

19 Sep '11

On Mon, 2011-09-19 at 16:34 +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/codecs/sn95031.c | 2 +- > 1 files changed, 1 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c > index b4f1cb4..29945b0 100644 > --- a/sound/soc/codecs/sn95031.c > +++ b/sound/soc/codecs/sn95031.c > @@ -658,7 +658,7 @@ static int sn95031_pcm_spkr_mute(struct snd_soc_dai *dai, int mute) > return 0; > } > > -int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, > +static int sn95031_pcm_hw_params(struct snd_pcm_substream *substream, > struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) > { > unsigned int format, rate; Acked-by: Vinod Koul <vinod.koul(a)linux.intel.com> -- ~Vinod

2 1

[alsa-devel] [PATCH 1/2] ASoC: Initial WM8996 headphone impedance measurement support
by Mark Brown 19 Sep '11

19 Sep '11

The WM8996 can measure the impedance of accessories connected to the headphone output. Implement initial support for this, measuring the left channel impedance when an accessory is detected and using this to distinguish between a line load and a headphone load. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 137 ++++++++++++++++++++++++++++++++++++-------- 1 files changed, 112 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8e9d7f7..a6168c5 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2350,12 +2350,94 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, /* Enable interrupts and we're off */ snd_soc_update_bits(codec, WM8996_INTERRUPT_STATUS_2_MASK, - WM8996_IM_MICD_EINT, 0); + WM8996_IM_MICD_EINT | WM8996_HP_DONE_EINT, 0); return 0; } EXPORT_SYMBOL_GPL(wm8996_detect); +static void wm8996_hpdet_irq(struct snd_soc_codec *codec) +{ + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int val, reg, report; + + /* Assume headphone in error conditions; we need to report + * something or we stall our state machine. + */ + report = SND_JACK_HEADPHONE; + + reg = snd_soc_read(codec, WM8996_HEADPHONE_DETECT_2); + if (reg < 0) { + dev_err(codec->dev, "Failed to read HPDET status\n"); + goto out; + } + + if (!(reg & WM8996_HP_DONE)) { + dev_err(codec->dev, "Got HPDET IRQ but HPDET is busy\n"); + goto out; + } + + val = reg & WM8996_HP_LVL_MASK; + + dev_dbg(codec->dev, "HPDET measured %d ohms\n", val); + + /* If we've got high enough impedence then report as line, + * otherwise assume headphone. + */ + if (val >= 126) + report = SND_JACK_LINEOUT; + else + report = SND_JACK_HEADPHONE; + +out: + if (wm8996->jack_mic) + report |= SND_JACK_MICROPHONE; + + snd_soc_jack_report(wm8996->jack, report, + SND_JACK_LINEOUT | SND_JACK_HEADSET); + + wm8996->detecting = false; + + /* If the output isn't running re-clamp it */ + if (!(snd_soc_read(codec, WM8996_POWER_MANAGEMENT_1) & + (WM8996_HPOUT1L_ENA | WM8996_HPOUT1R_RMV_SHORT))) + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, 0); + + /* Go back to looking at the microphone */ + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, + WM8996_MICD_ENA); + + snd_soc_dapm_disable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); +} + +static void wm8996_hpdet_start(struct snd_soc_codec *codec) +{ + /* Unclamp the output, we can't measure while we're shorting it */ + snd_soc_update_bits(codec, WM8996_ANALOGUE_HP_1, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT, + WM8996_HPOUT1L_RMV_SHORT | + WM8996_HPOUT1R_RMV_SHORT); + + /* We need bandgap for HPDET */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "Bandgap"); + snd_soc_dapm_sync(&codec->dapm); + + /* Go into headphone detect left mode */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_ENA, 0); + snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_1, + WM8996_JD_MODE_MASK, 1); + + /* Trigger a measurement */ + snd_soc_update_bits(codec, WM8996_HEADPHONE_DETECT_1, + WM8996_HP_POLL, WM8996_HP_POLL); +} + static void wm8996_micd(struct snd_soc_codec *codec) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); @@ -2376,28 +2458,36 @@ static void wm8996_micd(struct snd_soc_codec *codec) wm8996->jack_mic = false; wm8996->detecting = true; snd_soc_jack_report(wm8996->jack, 0, - SND_JACK_HEADSET | SND_JACK_BTN_0); + SND_JACK_LINEOUT | SND_JACK_HEADSET | + SND_JACK_BTN_0); + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, WM8996_MICD_RATE_MASK); return; } - /* If the measurement is very high we've got a microphone but - * do a little debounce to account for mechanical issues. + /* If the measurement is very high we've got a microphone, + * either we just detected one or if we already reported then + * we've got a button release event. */ if (val & 0x400) { - dev_dbg(codec->dev, "Microphone detected\n"); - snd_soc_jack_report(wm8996->jack, SND_JACK_HEADSET, - SND_JACK_HEADSET | SND_JACK_BTN_0); - wm8996->jack_mic = true; - wm8996->detecting = false; - - /* Increase poll rate to give better responsiveness - * for buttons */ - snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, - WM8996_MICD_RATE_MASK, - 5 << WM8996_MICD_RATE_SHIFT); + if (wm8996->detecting) { + dev_dbg(codec->dev, "Microphone detected\n"); + wm8996->jack_mic = true; + wm8996_hpdet_start(codec); + + /* Increase poll rate to give better responsiveness + * for buttons */ + snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, + WM8996_MICD_RATE_MASK, + 5 << WM8996_MICD_RATE_SHIFT); + } else { + dev_dbg(codec->dev, "Mic button up\n"); + snd_soc_jack_report(wm8996->jack, 0, SND_JACK_BTN_0); + } + + return; } /* If we detected a lower impedence during initial startup @@ -2429,15 +2519,11 @@ static void wm8996_micd(struct snd_soc_codec *codec) if (val & 0x3fc) { if (wm8996->jack_mic) { dev_dbg(codec->dev, "Mic button detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADSET | SND_JACK_BTN_0, - SND_JACK_HEADSET | SND_JACK_BTN_0); - } else { - dev_dbg(codec->dev, "Headphone detected\n"); - snd_soc_jack_report(wm8996->jack, - SND_JACK_HEADPHONE, - SND_JACK_HEADSET | + snd_soc_jack_report(wm8996->jack, SND_JACK_BTN_0, SND_JACK_BTN_0); + } else if (wm8996->detecting) { + dev_dbg(codec->dev, "Headphone detected\n"); + wm8996_hpdet_start(codec); /* Increase the detection rate a bit for * responsiveness. @@ -2445,8 +2531,6 @@ static void wm8996_micd(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8996_MIC_DETECT_1, WM8996_MICD_RATE_MASK, 7 << WM8996_MICD_RATE_SHIFT); - - wm8996->detecting = false; } } } @@ -2486,6 +2570,9 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); + if (irq_val & WM8996_HP_DONE_EINT) + wm8996_hpdet_irq(codec); + return IRQ_HANDLED; } -- 1.7.6.3

2 2

[alsa-devel] [PATCH 4/4] src/pcm/pcm_multi.c: add missing free
by Julia Lawall 18 Sep '11

18 Sep '11

From: Julia Lawall <julia(a)diku.dk> Something that is allocated using calloc is not freed on an error path. Signed-off-by: Julia Lawall <julia(a)diku.dk> Signed-off-by: Suman Saha <sumsaha(a)gmail.com> --- Compile-tested only. src/pcm/pcm_multi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/src/pcm/pcm_multi.c b/src/pcm/pcm_multi.c index 68f2d68..6b39c7a 100644 --- a/src/pcm/pcm_multi.c +++ b/src/pcm/pcm_multi.c @@ -886,6 +886,8 @@ int snd_pcm_multi_open(snd_pcm_t **pcmp, const char *name, err = snd_pcm_new(&pcm, SND_PCM_TYPE_MULTI, name, stream, multi->slaves[0].pcm->mode); if (err < 0) { + free(multi->slaves); + free(multi->channels); free(multi); return err; }

1 0

[alsa-devel] [PATCH 3/4] src/pcm/pcm_ladspa.c: add missing free
by Julia Lawall 18 Sep '11

18 Sep '11

From: Julia Lawall <julia(a)diku.dk> Something that is allocated using calloc is not freed on some error paths. Signed-off-by: Julia Lawall <julia(a)diku.dk> Signed-off-by: Suman Saha <sumsaha(a)gmail.com> --- Compile-tested only. src/pcm/pcm_ladspa.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index c413c10..84ebaa5 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -750,8 +750,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads if (instance->input.data == NULL || instance->input.m_data == NULL || instance->output.data == NULL || - instance->output.m_data == NULL) + instance->output.m_data == NULL) { + free(pchannels); return -ENOMEM; + } for (idx = 0; idx < instance->input.channels.size; idx++) { chn = instance->output.channels.array[idx]; if (pchannels[chn] == NULL && chn < ichannels) { @@ -761,8 +763,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->input.data[idx] = pchannels[chn]; if (instance->input.data[idx] == NULL) { instance->input.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 0); - if (instance->input.data[idx] == NULL) + if (instance->input.data[idx] == NULL) { + free(pchannels); return -ENOMEM; + } } } for (idx = 0; idx < instance->output.channels.size; idx++) { @@ -770,8 +774,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads /* FIXME/OPTIMIZE: check if we can remove double alloc */ /* if LADSPA plugin has no broken inplace */ instance->output.data[idx] = malloc(sizeof(LADSPA_Data) * ladspa->allocated); - if (instance->output.data[idx] == NULL) + if (instance->output.data[idx] == NULL) { + free(pchannels); return -ENOMEM; + } pchannels[chn] = instance->output.m_data[idx] = instance->output.data[idx]; } } @@ -793,8 +799,10 @@ static int snd_pcm_ladspa_allocate_memory(snd_pcm_t *pcm, snd_pcm_ladspa_t *lads instance->output.data[idx] = NULL; } else { instance->output.data[idx] = snd_pcm_ladspa_allocate_zero(ladspa, 1); - if (instance->output.data[idx] == NULL) + if (instance->output.data[idx] == NULL) { + free(pchannels); return -ENOMEM; + } } } }

1 0

[alsa-devel] [PATCH 1/1] sound/usb: Added support for Roland UM-ONE midi-usb interface.
by Daniele Guerrieri 18 Sep '11

18 Sep '11

Hi all, this is my first post to this list. Recently i bought a roland UM-ONE midi-usb interface which i thought worked well with alsa, but it didn't. So i got from lsusb id/vid and added them to sound/usb/quirks-table.h; it seems to work well now with snd-usb-audio, tested on 2 machines. Below there is the patch output for my changes. As a master branch i cloned git://github.com/torvalds/linux.git ----output of git format-patch ----- Roland UM-ONE midi usb interface differs from Roland UM-1. Signed-off-by: Daniele Guerrieri <d.guerrieri(a)gmail.com> --- sound/usb/quirks-table.h | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index a42e3ef..f041b2e 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -1192,6 +1192,20 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { + /* Added support for Roland UM-ONE which differs from UM-1 */ + USB_DEVICE(0x0582, 0x012a), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + /* .vendor_name = "ROLAND", */ + /* .product_name = "UM-ONE", */ + .ifnum = 0, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0003 + } + } +}, +{ /* has ID 0x006b when not in "Advanced Driver" mode */ USB_DEVICE_VENDOR_SPEC(0x0582, 0x006a), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- 1.7.6

2 2

[PATCH 2/2] saa7134: introduce "std" module parameter to force video std
by Stas Sergeev 18 Sep '11

18 Sep '11

--- drivers/media/video/saa7134/saa7134-video.c | 39 ++++++++++++++++++++++++-- drivers/media/video/saa7134/saa7134.h | 1 + 2 files changed, 37 insertions(+), 3 deletions(-) diff --git a/drivers/media/video/saa7134/saa7134-video.c b/drivers/media/video/saa7134/saa7134-video.c index 776ba2d..04ac7de 100644 --- a/drivers/media/video/saa7134/saa7134-video.c +++ b/drivers/media/video/saa7134/saa7134-video.c @@ -40,12 +40,15 @@ static unsigned int noninterlaced; /* 0 */ static unsigned int gbufsize = 720*576*4; static unsigned int gbufsize_max = 720*576*4; static char secam[] = "--"; +static char std[16] = "--"; module_param(video_debug, int, 0644); MODULE_PARM_DESC(video_debug,"enable debug messages [video]"); module_param(gbuffers, int, 0444); MODULE_PARM_DESC(gbuffers,"number of capture buffers, range 2-32"); module_param(noninterlaced, int, 0644); MODULE_PARM_DESC(noninterlaced,"capture non interlaced video"); +module_param_string(std, std, sizeof(std), 0644); +MODULE_PARM_DESC(secam, "force TV standard, either PAL, SECAM or NTSC"); module_param_string(secam, secam, sizeof(secam), 0644); MODULE_PARM_DESC(secam, "force SECAM variant, either DK,L or Lc"); @@ -1847,14 +1850,20 @@ int saa7134_s_std_internal(struct saa7134_dev *dev, struct saa7134_fh *fh, v4l2_ return -EBUSY; } - for (i = 0; i < TVNORMS; i++) + for (i = 0; i < TVNORMS; i++) { + if (!tvnorms[i].enabled) + continue; if (*id == tvnorms[i].id) break; + } if (i == TVNORMS) - for (i = 0; i < TVNORMS; i++) + for (i = 0; i < TVNORMS; i++) { + if (!tvnorms[i].enabled) + continue; if (*id & tvnorms[i].id) break; + } if (i == TVNORMS) return -EINVAL; @@ -1871,6 +1880,8 @@ int saa7134_s_std_internal(struct saa7134_dev *dev, struct saa7134_fh *fh, v4l2_ fixup = V4L2_STD_SECAM; } for (i = 0; i < TVNORMS; i++) { + if (!tvnorms[i].enabled) + continue; if (fixup == tvnorms[i].id) break; } @@ -2579,8 +2590,30 @@ int saa7134_videoport_init(struct saa7134_dev *dev) int saa7134_video_init2(struct saa7134_dev *dev) { + int i, idx = -1; + for (i = 0; i < TVNORMS; i++) { + int enable = (std[0] == '-' || strncasecmp(std, + tvnorms[i].name, strlen(std)) == 0); + + if (secam[0] != '-') { + char buf[16] = "SECAM"; + if (strncasecmp(buf, tvnorms[i].name, + strlen(buf)) == 0) { + strcat(buf, "-"); + strcat(buf, secam); + if (strncasecmp(buf, tvnorms[i].name, + strlen(buf)) != 0) + enable = 0; + } + } + tvnorms[i].enabled = enable; + if (enable && idx == -1) + idx = i; + } + if (idx == -1) + return -EINVAL; /* init video hw */ - set_tvnorm(dev,&tvnorms[0]); + set_tvnorm(dev,&tvnorms[idx]); video_mux(dev,0); saa7134_tvaudio_setmute(dev); saa7134_tvaudio_setvolume(dev,dev->ctl_volume); diff --git a/drivers/media/video/saa7134/saa7134.h b/drivers/media/video/saa7134/saa7134.h index 28eb103..a0eddce 100644 --- a/drivers/media/video/saa7134/saa7134.h +++ b/drivers/media/video/saa7134/saa7134.h @@ -77,6 +77,7 @@ enum saa7134_video_out { struct saa7134_tvnorm { char *name; v4l2_std_id id; + int enabled; /* video decoder */ unsigned int sync_control; -- 1.7.6 --------------080905040006020600030205--

1 0

[PATCH 1/2] saa7134: fix automute
by Stas Sergeev 18 Sep '11

18 Sep '11

--- drivers/media/video/saa7134/saa7134-tvaudio.c | 7 +++++-- 1 files changed, 5 insertions(+), 2 deletions(-) diff --git a/drivers/media/video/saa7134/saa7134-tvaudio.c b/drivers/media/video/saa7134/saa7134-tvaudio.c index 57e646b..62a6287 100644 --- a/drivers/media/video/saa7134/saa7134-tvaudio.c +++ b/drivers/media/video/saa7134/saa7134-tvaudio.c @@ -332,7 +332,7 @@ static int tvaudio_checkcarrier(struct saa7134_dev *dev, struct mainscan *scan) { __s32 left,right,value; - if (audio_debug > 1) { + if (audio_debug > 1 && (dev->tvnorm->id & scan->std)) { int i; dprintk("debug %d:",scan->carr); for (i = -150; i <= 150; i += 30) { @@ -546,6 +546,7 @@ static int tvaudio_thread(void *data) dev->tvnorm->name, carrier/1000, carrier%1000, max1, max2); dev->last_carrier = carrier; + dev->automute = !(dev->thread.scan1 > 1); } else if (0 != dev->last_carrier) { /* no carrier -- try last detected one as fallback */ @@ -553,6 +554,7 @@ static int tvaudio_thread(void *data) dprintk("audio carrier scan failed, " "using %d.%03d MHz [last detected]\n", carrier/1000, carrier%1000); + dev->automute = 1; } else { /* no carrier + no fallback -- use default */ @@ -560,9 +562,9 @@ static int tvaudio_thread(void *data) dprintk("audio carrier scan failed, " "using %d.%03d MHz [default]\n", carrier/1000, carrier%1000); + dev->automute = 1; } tvaudio_setcarrier(dev,carrier,carrier); - dev->automute = 0; saa_andorb(SAA7134_STEREO_DAC_OUTPUT_SELECT, 0x30, 0x00); saa7134_tvaudio_setmute(dev); /* find the exact tv audio norm */ @@ -1020,6 +1022,7 @@ int saa7134_tvaudio_init2(struct saa7134_dev *dev) } dev->thread.thread = NULL; + dev->thread.scan1 = dev->thread.scan2 = 0; if (my_thread) { saa7134_tvaudio_init(dev); /* start tvaudio thread */ -- 1.7.6 --------------080905040006020600030205 Content-Type: text/x-patch; name="0002-saa7134-introduce-std-module-parameter-to-force-vide.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0002-saa7134-introduce-std-module-parameter-to-force-vide.pa"; filename*1="tch"

1 0

[alsa-devel] [PATCH 0/7] ASoC: ep93xx: convert to use snd_soc_register_card()
by Mika Westerberg 18 Sep '11

18 Sep '11

Here is a series of patches which convert ep93xx ASoC drivers to use snd_soc_register_card(). This has the advantage that the machine drivers get loaded automatically by udev. Also this gets rid of machine_is_xxx() from the machine drivers. Tested on Sim.One. The edb93xx and snappercl15 changes are compile tested only. Mika Westerberg (7): ASoC: ep93xx-pcm: add MODULE_ALIAS ASoC: simone: convert to use snd_soc_register_card() ASoC: edb93xx: convert to use snd_soc_register_card() ASoC: snappercl15: convert to use snd_soc_register_card() ARM: ep93xx: simone: register audio platform device ARM: ep93xx: edb93xx: register audio platform device ARM: ep93xx: snappercl15: register audio platform device arch/arm/mach-ep93xx/edb93xx.c | 6 +++ arch/arm/mach-ep93xx/simone.c | 13 +++++++- arch/arm/mach-ep93xx/snappercl15.c | 13 +++++++- sound/soc/ep93xx/edb93xx.c | 60 ++++++++++++++++++--------------- sound/soc/ep93xx/ep93xx-pcm.c | 1 + sound/soc/ep93xx/simone.c | 64 ++++++++++++++++++++---------------- sound/soc/ep93xx/snappercl15.c | 53 ++++++++++++++++++++--------- 7 files changed, 136 insertions(+), 74 deletions(-) -- 1.7.5.4

6 19

[alsa-devel] hda- Disable Sticky PCM
by Raymond Yau 18 Sep '11

18 Sep '11

2011/9/15 Takashi Iwai <tiwai(a)suse.de>: > At Thu, 15 Sep 2011 22:23:12 +0800, > Raymond Yau wrote: >> >> BTW, there is some strange behaviour of the non-sticky pcm >> implementation on ad1988 >> >> Refer to commit 0e7adbe263f89ea2ef15b5af5e80a812b2a85025 >> >> Disable sticky PCM stream assignment for AD codecs >> >> why hda_codec_cleanup_stream is called twice and this also occur with >> capture and playback > > Isn't it just the power-save? The suspend calls > hda_cleanup_all_streams(). > > That is, the first call is the normal one, and it doesn't clean up the > stream actually because of stickiness. The second call is the real > cleanup at power-saving. > > > Takashi > by adding prinkt statement in really cleanup steam() function, really_cleanup_stream is called twice so why is second call if the stream is already really cleanup ALSA hda_intel.c:1755: azx_pcm_prepare: bufsize=0x10000, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x6, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1412: hda_codec_setup_stream: NID=0xa, stream=0x5, channel=0, format=0x4011 ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x4 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x6 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x5 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0xa really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x3 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x4 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x6 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x5 really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0xa really cleanup steam ALSA hda_codec.c:1475: hda_codec_cleanup_stream: NID=0x3 really cleanup steam

2 1

[alsa-devel] Noise on snd-hda-intel
by Sune Mølgaard 18 Sep '11

18 Sep '11

Plugging almost any headset into the front headset jacks on my machine results in noise in the headphones as well as in Skype conversations and/or recording. Said noise is, however not head in the regular speakers (except when playing back recordings, of course. lspci -vvnn gives 00:1b.0 Audio device [0403]: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller [8086:3a3e] Subsystem: ASUSTeK Computer Inc. Device [1043:82ea] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 256 bytes Interrupt: pin A routed to IRQ 70 Region 0: Memory at f7ff8000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel and aplay -l gives **** List of PLAYBACK Hardware Devices **** card 0: Device [USB Multimedia Audio Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 I have tried with no modules options as well as options snd_hda_intel model=6stack and options snd_hda_intel model=6stack-dig Best regards, Sune Mølgaard -- You buy Windows, and you put it on your machine-you can't do a whole lot with it. - Bob Muglia, Microsoft Senior Vice President

4 14

[alsa-devel] [PATCH] [v2] ASoC: support all possible sample rates in the WM8776 driver
by Timur Tabi 17 Sep '11

17 Sep '11

The .set_sysclk() function in ASoC codec drivers can be used to tell the driver what the frequency of master clock (mclk) is. Some codecs use a divider on this clock to determine the sample rate. The WM8776 is one such codec. So instead of hard-coding a list of specific sample rates supported, these codec drivers can specify SNDRV_PCM_RATE_CONTINUOUS instead, clamping the range to the absolute limits of the hardware. Then the .hw_params() function can use the mclk rate to determine whether any requested rate is actually supported. Although the WM8776 driver includes a .set_sysclk function, it was also hard-coding the list of supported sample rates. We change the hard-coded list to a range within the capabilities of the WM8776 itself. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/wm8776.c | 13 ++++++------- 1 files changed, 6 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ad6f0fa..ca8a593 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -321,11 +321,6 @@ static int wm8776_set_bias_level(struct snd_soc_codec *codec, return 0; } -#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ - SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ - SNDRV_PCM_RATE_96000) - - #define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) @@ -350,7 +345,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 192000, .formats = WM8776_FORMATS, }, .ops = &wm8776_dac_ops, @@ -362,7 +359,9 @@ static struct snd_soc_dai_driver wm8776_dai[] = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = WM8776_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 32000, + .rate_max = 96000, .formats = WM8776_FORMATS, }, .ops = &wm8776_adc_ops, -- 1.7.3.4

4 19

[alsa-devel] [PATCH 1/3] ASoC: support sample sizes properly in the WM8776 codec driver
by Timur Tabi 17 Sep '11

17 Sep '11

Use snd_pcm_format_width() to determine the sample size, instead of checking specify sample formats and assuming that those are the only valid format. This change adds support for big-endian architectures (which use the _BE formats) and the packed 24-bit format (SNDRV_PCM_FORMAT_S24_3xE). Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/wm8776.c | 25 +++++++++++++------------ 1 files changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 8e7953b..ad6f0fa 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -215,8 +215,6 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, int ratio_shift, master; int i; - iface = 0; - switch (dai->driver->id) { case WM8776_DAI_DAC: iface_reg = WM8776_DACIFCTRL; @@ -232,20 +230,23 @@ static int wm8776_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* Set word length */ - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S16_LE: + i = snd_pcm_format_width(params_format(params)); + switch (i) { + case 16: + iface = 0; + case 20: + iface = 0x10; break; - case SNDRV_PCM_FORMAT_S20_3LE: - iface |= 0x10; + case 24: + iface = 0x20; break; - case SNDRV_PCM_FORMAT_S24_LE: - iface |= 0x20; - break; - case SNDRV_PCM_FORMAT_S32_LE: - iface |= 0x30; + case 32: + iface = 0x30; break; + default: + dev_err(codec->dev, "Unsupported sample size: %i\n", i); + return -EINVAL; } /* Only need to set MCLK/LRCLK ratio if we're master */ -- 1.7.3.4

4 21

Re: [alsa-devel] Noise using pulseaudio with ymf724f
by Raymond Yau 17 Sep '11

17 Sep '11

2011/9/12 Clemens Ladisch <clemens(a)ladisch.de>: > Raymond Yau wrote: >> How accurate is the ymfpci timer ? > > The period interrupts are not accurate at all. The ymfpci hardware > internally uses fixed periods of 256 frames at 48 kHz; the driver > reports a period interrupt when the next hardware interrupt at or after > a period boundary occurs. The current position reported by the hardware > is the position at the time of the last hardware interrupt. > >> my ymf724f seem play quite well with alsa "hw" device but with bad >> noise with alsa-pulse plugin (e.g. aplay when not using default period >> times ) and it also occurrs on benshee on ubuntu 11.4 live cd > > It should be possible to increase accuracy by using the hardware period > parameters (48 kHz _and_ 256 frames/period). > > TODO: implement support for the NORESAMPLE flag. > > Is 256 frames/period in s16 stereo since the card support mono , stereo , U8 and S16 , What is the meaning of SNDRV_PCM_INFO_BLOCK_TRANSFER since the rewind_safeguard of PA server is 256 bytes ? Will this help since minimum period time of snd_ymfpci is 5.333ms but alsa-pulse plugin announce a lower value 128 bytes (period_bytes_min) to the alsa application ? PA server has #define ABSOLUTE_MIN_LATENCY (500) #define ABSOLUTE_MAX_LATENCY (10*PA_USEC_PER_SEC)

2 4

[alsa-devel] [linux-next] ASoC: sound/soc/samsung/neo1973_wm8753.c build failure
by Axel Lin 16 Sep '11

16 Sep '11

Hi, I got below build error: ( on linux-next 20110831 ) CC sound/soc/samsung/neo1973_wm8753.o sound/soc/samsung/neo1973_wm8753.c: In function 'neo1973_wm8753_init': sound/soc/samsung/neo1973_wm8753.c:325: error: implicit declaration of function 'machine_is_neo1973_gta01' make[3]: *** [sound/soc/samsung/neo1973_wm8753.o] Error 1 make[2]: *** [sound/soc/samsung] Error 2 make[1]: *** [sound/soc] Error 2 make: *** [sound] Error 2 I think the root cause is below commit: commit 6f82f4db "ARM: Update (and cut down) mach-types" removes neo1973_gta01 entry. I'm not sure if we should add back the neo1973_gta01 entry to mach-types or remove all machine_is_neo1973_gta01() calls in neo1973_wm8753.c. Comments? Regards, Axel

5 9

[alsa-devel] [PATCH] ASoC: WM8996 only needs bandgap for analogue functionality
by Mark Brown 16 Sep '11

16 Sep '11

Rather than managing the bandgap in the bias level control use a supply widget as we only actually need to enable it for analogue paths, not fully digital ones. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 37 +++++++++++++++++++++++++++++-------- 1 files changed, 29 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 5ca9d76..93fbb5b 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -666,6 +666,25 @@ SOC_SINGLE_TLV("DSP2 EQ B5 Volume", WM8996_DSP2_RX_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static int bg_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(2); + break; + default: + BUG(); + ret = -EINVAL; + } + + return ret; +} + static int cp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -999,7 +1018,8 @@ SND_SOC_DAPM_SUPPLY_S("AIFCLK", 2, WM8996_CLOCKING_1, 2, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Charge Pump", 2, WM8996_CHARGE_PUMP_1, 15, 0, cp_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), - +SND_SOC_DAPM_SUPPLY("Bandgap", WM8996_POWER_MANAGEMENT_1, WM8996_BG_ENA, 0, + bg_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("LDO2", WM8996_POWER_MANAGEMENT_2, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("MICB1 Audio", WM8996_MICBIAS_1, 4, 1, NULL, 0), SND_SOC_DAPM_SUPPLY("MICB2 Audio", WM8996_MICBIAS_2, 4, 1, NULL, 0), @@ -1159,18 +1179,22 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "MICB1", NULL, "LDO2" }, { "MICB1", NULL, "MICB1 Audio" }, + { "MICB1", NULL, "Bandgap" }, { "MICB2", NULL, "LDO2" }, { "MICB2", NULL, "MICB2 Audio" }, + { "MICB2", NULL, "Bandgap" }, { "IN1L PGA", NULL, "IN2LN" }, { "IN1L PGA", NULL, "IN2LP" }, { "IN1L PGA", NULL, "IN1LN" }, { "IN1L PGA", NULL, "IN1LP" }, + { "IN1L PGA", NULL, "Bandgap" }, { "IN1R PGA", NULL, "IN2RN" }, { "IN1R PGA", NULL, "IN2RP" }, { "IN1R PGA", NULL, "IN1RN" }, { "IN1R PGA", NULL, "IN1RP" }, + { "IN1R PGA", NULL, "Bandgap" }, { "ADCL", NULL, "IN1L PGA" }, @@ -1304,6 +1328,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "DAC2R", NULL, "DAC2R Mixer" }, { "HPOUT2L PGA", NULL, "Charge Pump" }, + { "HPOUT2L PGA", NULL, "Bandgap" }, { "HPOUT2L PGA", NULL, "DAC2L" }, { "HPOUT2L_DLY", NULL, "HPOUT2L PGA" }, { "HPOUT2L_DCS", NULL, "HPOUT2L_DLY" }, @@ -1311,6 +1336,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2L_RMV_SHORT", NULL, "HPOUT2L_OUTP" }, { "HPOUT2R PGA", NULL, "Charge Pump" }, + { "HPOUT2R PGA", NULL, "Bandgap" }, { "HPOUT2R PGA", NULL, "DAC2R" }, { "HPOUT2R_DLY", NULL, "HPOUT2R PGA" }, { "HPOUT2R_DCS", NULL, "HPOUT2R_DLY" }, @@ -1318,6 +1344,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT2R_RMV_SHORT", NULL, "HPOUT2R_OUTP" }, { "HPOUT1L PGA", NULL, "Charge Pump" }, + { "HPOUT1L PGA", NULL, "Bandgap" }, { "HPOUT1L PGA", NULL, "DAC1L" }, { "HPOUT1L_DLY", NULL, "HPOUT1L PGA" }, { "HPOUT1L_DCS", NULL, "HPOUT1L_DLY" }, @@ -1325,6 +1352,7 @@ static const struct snd_soc_dapm_route wm8996_dapm_routes[] = { { "HPOUT1L_RMV_SHORT", NULL, "HPOUT1L_OUTP" }, { "HPOUT1R PGA", NULL, "Charge Pump" }, + { "HPOUT1R PGA", NULL, "Bandgap" }, { "HPOUT1R PGA", NULL, "DAC1R" }, { "HPOUT1R_DLY", NULL, "HPOUT1R PGA" }, { "HPOUT1R_DCS", NULL, "HPOUT1R_DLY" }, @@ -1643,14 +1671,7 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - break; - case SND_SOC_BIAS_PREPARE: - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) { - snd_soc_update_bits(codec, WM8996_POWER_MANAGEMENT_1, - WM8996_BG_ENA, WM8996_BG_ENA); - msleep(2); - } break; case SND_SOC_BIAS_STANDBY: -- 1.7.6.3

1 0

[alsa-devel] [PATCH] ASoC: Add support for Cirrus Logic CS42L73 Codec
by Brian Austin 16 Sep '11

16 Sep '11

--- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs42l73.c | 1372 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l73.h | 225 ++++++++ 4 files changed, 1603 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs42l73.c create mode 100644 sound/soc/codecs/cs42l73.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 71b46c8..dca1183 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -28,6 +28,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 @@ -173,6 +174,9 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L73 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 70c1769..bdbc58d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -15,6 +15,7 @@ snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o @@ -113,6 +114,7 @@ obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c new file mode 100644 index 0000000..a2b9ce3 --- /dev/null +++ b/sound/soc/codecs/cs42l73.c @@ -0,0 +1,1372 @@ +/* + * cs42l73.c -- CS42L73 ALSA Soc Audio driver + * + * Copyright 2011 Cirrus Logic, Inc. + * + * Authors: Georgi Vlaev, Nucleus Systems Ltd, <office(a)nucleusys.com> + * Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/gpio.h> + +#include "cs42l73.h" + +struct sp_config { + u8 spc, mmcc, spfs; + u32 srate; +}; + +struct cs42l73_private { + enum snd_soc_control_type control_type; + void *control_data; + u8 reg_cache[CS42L73_CACHEREGNUM]; + u32 sysclk; /* external MCLK */ + u8 mclksel; /* MCLKx */ + u32 mclk; /* internal MCLK */ + struct sp_config config[3]; +}; + +static const u8 cs42l73_reg[CS42L73_CACHEREGNUM] = { +/* 0*/ 0x00, 0x42, 0xA7, 0x30, +/* 4*/ 0x00, 0x00, 0xF1, 0xDF, +/* 8*/ 0x3F, 0x57, 0x53, 0x00, +/* C*/ 0x00, 0x15, 0x00, 0x15, +/*10*/ 0x00, 0x15, 0x00, 0x06, +/*14*/ 0x00, 0x00, 0x00, 0x00, +/*18*/ 0x00, 0x00, 0x00, 0x00, +/*1C*/ 0x00, 0x00, 0x00, 0x00, +/*20*/ 0x00, 0x00, 0x00, 0x00, +/*24*/ 0x00, 0x00, 0x00, 0x7F, +/*28*/ 0x00, 0x00, 0x3F, 0x00, +/*2C*/ 0x00, 0x3F, 0x00, 0x00, +/*30*/ 0x3F, 0x00, 0x00, 0x00, +/*34*/ 0x18, 0x3F, 0x3F, 0x3F, +/*38*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*3C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*40*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*44*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*48*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*4C*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*50*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*54*/ 0x3F, 0xAA, 0x3F, 0x3F, +/*58*/ 0x3F, 0x3F, 0x3F, 0x3F, +/*5C*/ 0x3F, 0x3F, 0x00, 0x00, +/*60*/ 0x00, 0x00 +}; + +/* + CS42L73 I2C read/write. + 7 bit address, 8 bit data +*/ +static inline int cs42l73_read_reg_cache(struct snd_soc_codec *codec, u_int reg) +{ + u8 *cache = codec->reg_cache; + + return reg > CS42L73_CACHEREGNUM ? -EINVAL : cache[reg]; +} + +static inline void cs42l73_write_reg_cache(struct snd_soc_codec *codec, + u_int reg, u_int val) +{ + u8 *cache = codec->reg_cache; + + if (reg > CS42L73_CACHEREGNUM) + return; + + cache[reg] = val & 0xff; +} + +int cs42l73_write(struct snd_soc_codec *codec, unsigned reg, u_int val) +{ + u8 data[2]; + + if (reg > CS42L73_CACHEREGNUM) + return -EINVAL; + + cs42l73_write_reg_cache(codec, reg, val); + + data[0] = reg & 0x7f; /* reg address */ + data[1] = val & 0xff; /* reg value */ + + dev_dbg(codec->dev, "%s: reg 0x%x = %02x (%d)\n", + __FUNCTION__, reg, val, val); + + if (codec->hw_write(codec->control_data, data, 2) != 2) + return -EIO; + + return 0; +} + +unsigned int cs42l73_read(struct snd_soc_codec *codec, u_int reg) +{ + u8 *cache = codec->reg_cache; + + dev_dbg(codec->dev, "%s: reg 0x%x = %02x (%d)\n", + __FUNCTION__, reg, cache[reg], cache[reg]); + + return cache[reg]; +} + +int cs42l73_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mmax = (max > min) ? max:min; + unsigned int mask = (1 << fls(mmax)) - 1; + + ucontrol->value.integer.value[0] = + ((cs42l73_read(codec, reg) >> shift) - min) & mask; + if (shift != rshift) + ucontrol->value.integer.value[1] = + ((cs42l73_read(codec, reg) >> rshift) - min) & mask; + + return 0; +} + +int cs42l73_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int rshift = mc->rshift; + int max = mc->max; + int min = mc->min; + int mmax = (max > min) ? max:min; + unsigned int mask = (1 << fls(mmax)) - 1; + unsigned short val, val2, val_mask; + + val = ((ucontrol->value.integer.value[0] + min) & mask); + + val_mask = mask << shift; + val = val << shift; + if (shift != rshift) { + val2 = ((ucontrol->value.integer.value[1] + min) & mask); + val_mask |= mask << rshift; + val |= val2 << rshift; + } + return snd_soc_update_bits(codec, reg, val_mask, val); +} + +int cs42l73_info_volsw_2r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + int max = mc->max; + + if (max == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = max; + return 0; +} + +int cs42l73_get_volsw_2r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + int max = mc->max; + int min = mc->min; + int mmax = (max > min) ? max:min; + unsigned int mask = (1 << fls(mmax)) - 1; + int val, val2; + + val = cs42l73_read(codec, reg); + val2 = cs42l73_read(codec, reg2); + ucontrol->value.integer.value[0] = (val - min) & mask; + ucontrol->value.integer.value[1] = (val2 - min) & mask; + return 0; +} + +int cs42l73_put_volsw_2r(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + + unsigned int reg = mc->reg; + unsigned int reg2 = mc->rreg; + int max = mc->max; + int min = mc->min; + int mmax = (max > min) ? max:min; + unsigned int mask = (1 << fls(mmax)) - 1; + int err; + unsigned short val, val2; + + val = (ucontrol->value.integer.value[0] + min) & mask; + val2 = (ucontrol->value.integer.value[1] + min) & mask; + + if ((err = snd_soc_update_bits(codec, reg, mask, val)) < 0) + return err; + + return snd_soc_update_bits(codec, reg2, mask, val2); +} + +#define SOC_SINGLE_S8_C_TLV(xname, xreg, xshift, xmax, xmin, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = snd_soc_info_volsw, .get = cs42l73_get_volsw,\ + .put = cs42l73_put_volsw, .tlv.p = (tlv_array),\ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = xshift, .rshift = xshift, \ + .max = xmax, .min = xmin} } + +#define SOC_DOUBLE_R_S8_C_TLV(xname, xreg, xrreg, xmax, xmin, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = cs42l73_info_volsw_2r, \ + .get = cs42l73_get_volsw_2r, .put = cs42l73_put_volsw_2r, \ + .tlv.p = (tlv_array), \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .rreg = xrreg, .max = xmax, .min = xmin} } + +/* + HP,LO Analog Volume TLV + -76dB ... -50 dB in 2dB steps + -50dB ... 12dB in 1dB steps +*/ +static const unsigned int hpaloa_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 13, TLV_DB_SCALE_ITEM(-7600, 200, 0), + 14,75, TLV_DB_SCALE_ITEM(-4900, 100, 0), +}; + +/* -102dB ... 12 dB in 0.5 dB steps */ +static DECLARE_TLV_DB_SCALE(hl_tlv, -10200, 50, 0); + +/* -96dB ... 12 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(ipd_tlv, -9600, 100, 0); + +/* -6dB ... 12 dB in 0.5 dB steps */ +static DECLARE_TLV_DB_SCALE(micpga_tlv, -600, 50, 0); + +/* + HL, ESL, SPK, Limiter Threshold/Cushion TLV + 0dB -12 dB in -3dB steps + -12dB -30dB in -6dB steps +*/ +static const unsigned int limiter_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), + 3, 7, TLV_DB_SCALE_ITEM(-1200, 300, 0), +}; + +/* + * Stereo Mixer Input Attenuation (regs 35h-54h) TLV + * Mono Mixer Input Attenuation (regs 56h-5Dh) + + -62dB ... 0dB in 1dB steps, < -62dB = mute +*/ +static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); + +/* Stereo Attenuation Group */ +#define SOC_DOUBLE_R_CS42L73_ATTN_GRP(xdest, xregl_start) \ + SOC_DOUBLE_R_TLV(xdest"-IP Attenuation Volume",\ + xregl_start + 0, xregl_start + 1, 0, 0x3F, 1, attn_tlv), \ + SOC_DOUBLE_R_TLV(xdest"-XSP Attenuation Volume",\ + xregl_start + 2, xregl_start + 3, 0, 0x3F, 1, attn_tlv), \ + SOC_DOUBLE_R_TLV(xdest"-ASP Attenuation Volume",\ + xregl_start + 4, xregl_start + 5, 0, 0x3F, 1, attn_tlv), \ + SOC_DOUBLE_R_TLV(xdest"-VSP Attenuation Volume",\ + xregl_start + 6, xregl_start + 7, 0, 0x3F, 1, attn_tlv) + +/* Mono Attenuation Group */ +#define SOC_SINGLE_CS42L73_ATTN_GRP(xdest, xreg_start) \ + SOC_SINGLE_TLV(xdest"-IP Mono Attenuation Volume",\ + xreg_start + 0, 0, 0x3F, 1, attn_tlv), \ + SOC_SINGLE_TLV(xdest"-XSP Mono Attenuation Volume",\ + xreg_start + 1, 0, 0x3F, 1, attn_tlv), \ + SOC_SINGLE_TLV(xdest"-ASP Mono Attenuation Volume",\ + xreg_start + 2, 0, 0x3F, 1, attn_tlv), \ + SOC_SINGLE_TLV(xdest"-VSP Mono Attenuation Volume",\ + xreg_start + 3, 0, 0x3F, 1, attn_tlv) + +/* Analog Input PGA Mux */ +static const char *cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; +static const char *cs42l73_pgab_text[] = { "Line B", "Mic 2" }; + +static const struct soc_enum pgaa_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, + ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); + +static const struct soc_enum pgab_enum = + SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, + ARRAY_SIZE (cs42l73_pgab_text), cs42l73_pgab_text); + +static const struct snd_kcontrol_new pgaa_mux = +SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); + +static const struct snd_kcontrol_new pgab_mux = +SOC_DAPM_ENUM("Right Analog Input Capture Mux", pgab_enum); + +/* NG */ +static const char *cs42l73_ng_delay_text[] = + { "50ms", "100ms", "150ms", "200ms" }; + +static const struct soc_enum ng_delay_enum = + SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, + ARRAY_SIZE (cs42l73_ng_delay_text), cs42l73_ng_delay_text); + +/* Mono Mixer Select*/ +static const char *cs42l73_mono_mixer_text[] = + { "Left", "Right", "Mono Mix"}; + +/* ESL-ASP, ESL-XSP, SPK-ASP, SPK-XSP Mono Mixer Selects */ +static const struct soc_enum mono_mixer_enum[] = +{ + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 6, + ARRAY_SIZE (cs42l73_mono_mixer_text), cs42l73_mono_mixer_text), + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 4, + ARRAY_SIZE (cs42l73_mono_mixer_text), cs42l73_mono_mixer_text), + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 2, + ARRAY_SIZE (cs42l73_mono_mixer_text), cs42l73_mono_mixer_text), + SOC_ENUM_SINGLE(CS42L73_MMIXCTL, 0, + ARRAY_SIZE (cs42l73_mono_mixer_text), cs42l73_mono_mixer_text), +}; + +static const char *cs42l73_ip_swap_text[] = + { "Stereo", "Mono A", "Mono B", "Swap A-B"}; + +static const struct soc_enum ip_swap_enum = + SOC_ENUM_SINGLE( CS42L73_MIOPC, 6, + ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); + +/* XSPOUT, VSPOUT Mixer output */ +static const char *cs42l73_spo_mixer_text[] = + { "Mono", "Stereo"}; + +static const struct soc_enum spo_mixer_enum[] = +{ + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, + ARRAY_SIZE (cs42l73_spo_mixer_text), cs42l73_spo_mixer_text), + SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, + ARRAY_SIZE (cs42l73_spo_mixer_text), cs42l73_spo_mixer_text), +}; + +static const struct snd_kcontrol_new cs42l73_snd_controls[] = { +/* + SOC_DOUBLE_R_S8_C (..., max, min) + min - min from CS datasheet + max - fls(max) - min + max from CS datasheet +*/ +/* Volume */ + SOC_DOUBLE_R_S8_C_TLV("Headphone Analog Playback Volume", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 0x4B, 0x41, hpaloa_tlv), + + SOC_DOUBLE_R_S8_C_TLV("LineOut Analog Playback Volume", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 0x4B, 0x41, hpaloa_tlv), + + SOC_DOUBLE_R_S8_C_TLV("Input PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 0x24, 0x34, micpga_tlv), + + SOC_DOUBLE_R("MIC Preamp Switch", CS42L73_MICAPREPGAAVOL, + CS42L73_MICBPREPGABVOL, 6, 1, 1), + + SOC_DOUBLE_R_S8_C_TLV("Input Path Digital Volume", CS42L73_IPADVOL, + CS42L73_IPBDVOL, 0x6C, 0xA0, ipd_tlv), + + SOC_DOUBLE_R_S8_C_TLV("HL Digital Playback Volume", + CS42L73_HLADVOL, CS42L73_HLBDVOL, 0xE4, 0x34, hl_tlv), + +/* Single select Volume */ + SOC_SINGLE_S8_C_TLV("Headphone A Analog Playback Volume", CS42L73_HPAAVOL, + 0, 0x4B, 0x41, hpaloa_tlv), + SOC_SINGLE_S8_C_TLV("Headphone B Analog Playback Volume", CS42L73_HPBAVOL, + 0, 0x4B, 0x41, hpaloa_tlv), + + SOC_SINGLE_S8_C_TLV("HL-A Digital Playback Volume", + 0, CS42L73_HLADVOL, 0xE4, 0x34, hl_tlv), + SOC_SINGLE_S8_C_TLV("HL-B Digital Playback Volume", + 0, CS42L73_HLBDVOL, 0xE4, 0x34, hl_tlv), + + SOC_SINGLE_S8_C_TLV("LineOut Analog A Playback Volume", CS42L73_LOAAVOL, + 0, 0x4B, 0x41, hpaloa_tlv), + SOC_SINGLE_S8_C_TLV("LineOut Analog B Playback Volume", CS42L73_LOBAVOL, + 0, 0x4B, 0x41, hpaloa_tlv), + + SOC_SINGLE_S8_C_TLV("MIC 1 PGA Analog Volume", CS42L73_MICAPREPGAAVOL, + 0, 0x24, 0x34, micpga_tlv), + SOC_SINGLE_S8_C_TLV("MIC 2 PGA Analog Volume", CS42L73_MICBPREPGABVOL, + 0, 0x24, 0x34, micpga_tlv), + + SOC_SINGLE_S8_C_TLV("Input Path A Digital Volume", CS42L73_IPADVOL, + 0, 0x6C, 0xA0, ipd_tlv), + SOC_SINGLE_S8_C_TLV("Input Path B Digital Volume", CS42L73_IPBDVOL, + 0, 0x6C, 0xA0, ipd_tlv), + + SOC_SINGLE_S8_C_TLV("Speakerphone Digital Playback Volume", CS42L73_SPKDVOL, + 0, 0xE4, 0x34, hl_tlv), + + SOC_SINGLE_S8_C_TLV("Ear Speaker Digital Playback Volume", CS42L73_ESLDVOL, + 0, 0xE4, 0x34, hl_tlv), + +/* Digital/Analog Mute */ + SOC_DOUBLE_R("Headphone Analog Playback Switch", CS42L73_HPAAVOL, + CS42L73_HPBAVOL, 7, 1, 1), + SOC_SINGLE("Headphone A Analog Playback Switch", CS42L73_HPAAVOL, 7, 1, 1), + SOC_SINGLE("Headphone B Analog Playback Switch", CS42L73_HPBAVOL, 7, 1, 1), + + SOC_DOUBLE_R("LineOut Analog Playback Switch", CS42L73_LOAAVOL, + CS42L73_LOBAVOL, 7, 1, 1), + SOC_SINGLE("LineOut A Analog Playback Switch", CS42L73_LOAAVOL, 7, 1, 1), + SOC_SINGLE("LineOut B Analog Playback Switch", CS42L73_LOBAVOL, 7, 1, 1), + + SOC_DOUBLE("Input Path Digital Switch", CS42L73_ADCIPC, 0, 4, 1, 1), + SOC_DOUBLE("HL Digital Playback Switch", CS42L73_PBDC, 0, + 1, 1, 1), + SOC_SINGLE("Speakerphone Digital Playback Switch", CS42L73_PBDC, 2, 1, + 1), + SOC_SINGLE("Ear Speaker Digital Playback Switch", CS42L73_PBDC, 3, 1, + 1), + + SOC_SINGLE("PGA Soft-Ramp Switch", CS42L73_MIOPC, 3, 1, 0), + SOC_SINGLE("Analog Zero Cross Switch", CS42L73_MIOPC, 2, 1, 0), + SOC_SINGLE("Digital Soft-Ramp Switch", CS42L73_MIOPC, 1, 1, 0), + SOC_SINGLE("Analog Output Soft-Ramp Switch", CS42L73_MIOPC, 0, 1, 0), + +/* ADC */ + SOC_DOUBLE("Invert ADC Signal Polarity Switch", CS42L73_ADCIPC, 1, 5, 1, + 0), + SOC_DOUBLE("ADC Boost Switch", CS42L73_ADCIPC, 2, 6, 1, 0), + + SOC_SINGLE("Charge Pump Frequency Volume", CS42L73_CPFCHC, 4, 15, 0), + +/* Headphone/LineOut (HL) Limiter */ + SOC_SINGLE("HL Limiter Attack Rate Volume", CS42L73_LIMARATEHL, 0, 0x3F, + 0), + SOC_SINGLE("HL Limiter Release Rate Volume", CS42L73_LIMRRATEHL, 0, + 0x3F, 0), + SOC_SINGLE("HL Limiter Switch", CS42L73_LIMRRATEHL, 7, 1, 0), + SOC_SINGLE("HL Limiter All Channels Switch", CS42L73_LIMRRATEHL, 6, 1, + 0), + + SOC_SINGLE_TLV("HL Limiter Max Threshold Volume", CS42L73_LMAXHL, 5, 7, + 1, + limiter_tlv), + SOC_SINGLE_TLV("HL Limiter Cushion Volume", CS42L73_LMAXHL, 2, 7, 1, + limiter_tlv), + +/* Speakerphone Limiter */ + SOC_SINGLE("SPK Limiter Attack Rate Volume", CS42L73_LIMARATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Release Rate Volume", CS42L73_LIMRRATESPK, 0, + 0x3F, 0), + SOC_SINGLE("SPK Limiter Switch", CS42L73_LIMRRATESPK, 7, 1, 0), + SOC_SINGLE("SPK Limiter All Channels Switch", CS42L73_LIMRRATESPK, 6, 1, + 0), + SOC_SINGLE_TLV("SPK Limiter Max Threshold Volume", CS42L73_LMAXSPK, 5, + 7, 1, + limiter_tlv), + SOC_SINGLE_TLV("SPK Limiter Cushion Volume", CS42L73_LMAXSPK, 2, 7, 1, + limiter_tlv), + +/* Earphone/Speakerphone LO Limiter */ + SOC_SINGLE("ESL Limiter Attack Rate Volume", CS42L73_LIMARATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Release Rate Volume", CS42L73_LIMRRATEESL, 0, + 0x3F, 0), + SOC_SINGLE("ESL Limiter Switch", CS42L73_LIMRRATEESL, 7, 1, 0), + SOC_SINGLE_TLV("ESL Limiter Max Threshold Volume", CS42L73_LMAXESL, 5, + 7, 1, + limiter_tlv), + SOC_SINGLE_TLV("ESL Limiter Cushion Volume", CS42L73_LMAXESL, 2, 7, 1, + limiter_tlv), + +/* ALC */ + + SOC_SINGLE("ALC Attack Rate Volume", CS42L73_ALCARATE, 0, 0x3F, 0), + SOC_SINGLE("ALC Release Rate Volume", CS42L73_ALCRRATE, 0, 0x3F, 0), + SOC_DOUBLE("ALC Switch", CS42L73_ALCARATE, 6, 7, 1, 0), + SOC_SINGLE_TLV("ALC Max Threshold Volume", CS42L73_ALCMINMAX, 5, 7, 1, + limiter_tlv), + SOC_SINGLE_TLV("ALC Min Threshold Volume", CS42L73_ALCMINMAX, 2, 7, 1, + limiter_tlv), + +/* Noise Gate */ + SOC_DOUBLE("NG Enable Switch", CS42L73_NGCAB, 6, 7, 1, 0), + SOC_SINGLE("NG Boost Switch", CS42L73_NGCAB, 5, 1, 0), + /* + NG Threshold depends on NG_BOOTSAB, which selects + between two threshold scales in decibels. + Set linear values for now .. + */ + SOC_SINGLE("NG Threshold", CS42L73_NGCAB, 2, 7, 0), + SOC_ENUM("NG Delay", ng_delay_enum), + +/* Digital IO Attenuation */ + SOC_DOUBLE_R_CS42L73_ATTN_GRP("XSP", CS42L73_XSPAIPAA), + SOC_DOUBLE_R_CS42L73_ATTN_GRP("ASP", CS42L73_ASPAIPAA), + SOC_DOUBLE_R_CS42L73_ATTN_GRP("VSP", CS42L73_VSPAIPAA), + +/* Output Attenuation */ + SOC_DOUBLE_R_CS42L73_ATTN_GRP("HL", CS42L73_HLAIPAA), + SOC_SINGLE_CS42L73_ATTN_GRP("SPK", CS42L73_SPKMIPMA), + SOC_SINGLE_CS42L73_ATTN_GRP("ESL", CS42L73_ESLMIPMA), + +/* Channel Swap Record Enum */ + SOC_ENUM("IP Digital Swap/Mono Select", ip_swap_enum), + +/* Mono Mixer Select*/ + SOC_ENUM("ESL-ASP Mono Mixer Select", mono_mixer_enum[0]), + SOC_ENUM("ESL-XSP Mono Mixer Select", mono_mixer_enum[1]), + SOC_ENUM("SPK-ASP Mono Mixer Select", mono_mixer_enum[2]), + SOC_ENUM("SPK-XSP Mono Mixer Select", mono_mixer_enum[3]), + +/* XSPOUT, VSPOUT Output Mixer Path Select */ + SOC_ENUM("VSP Output Mixer Select", spo_mixer_enum[0]), + SOC_ENUM("XSP Output Mixer Select", spo_mixer_enum[1]), + + + +}; + +static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { +/* Platform / Analog Inputs */ + SND_SOC_DAPM_INPUT("LINEINA"), + SND_SOC_DAPM_INPUT("LINEINB"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_MICBIAS("MIC1 Bias", CS42L73_PWRCTL2, 6, 1), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_MICBIAS("MIC2 Bias", CS42L73_PWRCTL2, 7, 1), + SND_SOC_DAPM_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), + +/* Stream */ + /* Digital Outputs*/ + SND_SOC_DAPM_AIF_OUT("XSPOUT", "XSP Capture", 0, CS42L73_PWRCTL2, 1, 1), + SND_SOC_DAPM_AIF_OUT("ASPOUT", "ASP Capture", 0, CS42L73_PWRCTL2, 3, 1), + SND_SOC_DAPM_AIF_OUT("VSPOUT", "VSP Capture", 0, CS42L73_PWRCTL2, 4, 1), + + /* Digital Inputs*/ + SND_SOC_DAPM_AIF_IN("XSPIN", "XSP Playback", 0, CS42L73_PWRCTL2, 0, 1), + SND_SOC_DAPM_AIF_IN("ASPIN", "ASP Playback", 0, CS42L73_PWRCTL2, 2, 1), + SND_SOC_DAPM_AIF_IN("VSPIN", "VSP Playback", 0, CS42L73_PWRCTL2, 4, 1), + + SND_SOC_DAPM_ADC("ADC Left", NULL, CS42L73_PWRCTL1, 5, 1), + SND_SOC_DAPM_ADC("ADC Right", NULL, CS42L73_PWRCTL1, 7, 1), + SND_SOC_DAPM_DAC("DAC Left", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC Right", NULL, SND_SOC_NOPM, 0, 0), + +/* Path */ + SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("PGA Right", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("PGA Mux Left", SND_SOC_NOPM, 0, 0, &pgaa_mux), + SND_SOC_DAPM_MUX("PGA Mux Right", SND_SOC_NOPM, 0, 0, &pgab_mux), +/* HP Output PGA */ + SND_SOC_DAPM_PGA("HP Amp Left", CS42L73_PWRCTL3, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("HP Amp Right", CS42L73_PWRCTL3, 0, 1, NULL, 0), +/* Line Output PGA */ + SND_SOC_DAPM_PGA("LO Amp Left", CS42L73_PWRCTL3, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("LO Amp Right", CS42L73_PWRCTL3, 1, 1, NULL, 0), +/* SPK Output PGA */ + SND_SOC_DAPM_PGA("SPK Amp", CS42L73_PWRCTL3, 2, 1, NULL, 0), +/* ESL Output PGA */ + SND_SOC_DAPM_PGA("EAR Amp", CS42L73_PWRCTL3, 3, 1, NULL, 0), +/* SPK LO PGA */ + SND_SOC_DAPM_PGA("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, NULL, 0), +/* Outputs */ + SND_SOC_DAPM_OUTPUT("HPOUTA"), + SND_SOC_DAPM_OUTPUT("HPOUTB"), + SND_SOC_DAPM_OUTPUT("LINEOUTA"), + SND_SOC_DAPM_OUTPUT("LINEOUTB"), + SND_SOC_DAPM_OUTPUT("EAROUT"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + SND_SOC_DAPM_OUTPUT("SPKLINEOUT"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* outputs */ + {"HPOUTA", NULL, "HP Amp Left"}, + {"HPOUTB", NULL, "HP Amp Right"}, + {"LINEOUTA", NULL, "LO Amp Left"}, + {"LINEOUTB", NULL, "LO Amp Right"}, + {"SPKOUT", NULL, "SPK Amp"}, + {"EAROUT", NULL, "EAR Amp"}, + {"SPKLINEOUT", NULL, "SPKLO Amp"}, + + {"HP Amp Left", "DAC", "DAC Left"}, + {"HP Amp Right", "DAC", "DAC Right"}, + {"LO Amp Left", "DAC", "DAC Left"}, + {"LO Amp Right", "DAC", "DAC Right"}, + {"SPK Amp", "DAC", "DAC Left"}, + {"SPKLO Amp", "DAC", "DAC Right"}, + {"EAR Amp", "DAC", "DAC Right"}, + + /* inputs */ + {"PGA Mux Left", NULL, "LINEINA"}, + {"PGA Mux Right", NULL, "LINEINB"}, + {"PGA Mux Left", NULL, "MIC1"}, + {"PGA Mux Right", NULL, "MIC2"}, + + {"PGA Left", NULL, "PGA Mux Left"}, + {"PGA Right", NULL, "PGA Mux Right"}, + {"ADC Left", "ADC", "PGA Left"}, + {"ADC Right", "ADC", "PGA Right"}, + + /* AIFx = [XSP,ASP,VSP] */ + {"XSPOUT", NULL, "ADC Left"}, + {"XSPOUT", NULL, "ADC Right"}, + {"DAC Left", NULL, "XSPIN"}, + {"DAC Right", NULL, "XSPIN"}, + + {"ASPOUT", NULL, "ADC Left"}, + {"ASPOUT", NULL, "ADC Right"}, + {"DAC Left", NULL, "ASPIN"}, + {"DAC Right", NULL, "ASPIN"}, + + {"VSPOUT", NULL, "ADC Left"}, + {"VSPOUT", NULL, "ADC Right"}, + {"DAC Left", NULL, "VSPIN"}, + {"DAC Right", NULL, "VSPIN"}, + +}; + +static int cs42l73_add_widgets(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_new_controls(dapm, cs42l73_dapm_widgets, + ARRAY_SIZE(cs42l73_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + + return 0; +} + +struct cs42l73_mclk_div { + u32 mclk; /* MCLK (MHz) */ + u32 srate; /* Sample rate (KHz) */ + u8 mmcc; /* x_MMCC[5:0] divider */ +}; + +struct cs42l73_mclk_div cs42l73_mclk_coeffs[] = { + /* MCLK, Sample Rate, xMMCC[5:0] */ + {5644800, 11025, 0x30}, + {5644800, 22050, 0x20}, + {5644800, 44100, 0x10}, + + {6000000, 8000, 0x39}, + {6000000, 11025, 0x33}, + {6000000, 12000, 0x31}, + {6000000, 16000, 0x29}, + {6000000, 22050, 0x23}, + {6000000, 24000, 0x21}, + {6000000, 32000, 0x19}, + {6000000, 44100, 0x13}, + {6000000, 48000, 0x11}, + + {6144000, 8000, 0x38}, + {6144000, 12000, 0x30}, + {6144000, 16000, 0x28}, + {6144000, 24000, 0x20}, + {6144000, 32000, 0x18}, + {6144000, 48000, 0x10}, + + {6500000, 8000, 0x3C}, + {6500000, 11025, 0x35}, + {6500000, 12000, 0x34}, + {6500000, 16000, 0x2C}, + {6500000, 22050, 0x25}, + {6500000, 24000, 0x24}, + {6500000, 32000, 0x1C}, + {6500000, 44100, 0x15}, + {6500000, 48000, 0x14}, + + {6400000, 8000, 0x3E}, + {6400000, 11025, 0x37}, + {6400000, 12000, 0x36}, + {6400000, 16000, 0x2E}, + {6400000, 22050, 0x27}, + {6400000, 24000, 0x26}, + {6400000, 32000, 0x1E}, + {6400000, 44100, 0x17}, + {6400000, 48000, 0x16}, +}; + +struct cs42l73_mcklx_div { + u32 mclkx; /* MCLK1/2 (MHz) */ + u8 ratio; /* Required Divide Ratio */ + u8 mclkdiv; /* MCLKDIV[2:0] */ +}; + +struct cs42l73_mcklx_div cs42l73_mclkx_coeffs[] = { + {5644800, 1, 0}, /* 5644800 */ + {6000000, 1, 0}, /* 6000000 */ + {6144000, 1, 0}, /* 6144000 */ + {11289600, 2, 2}, /* 5644800 */ + {12288000, 2, 2}, /* 6144000 */ + {12000000, 2, 2}, /* 6000000 */ + {13000000, 2, 2}, /* 6500000 */ + {19200000, 3, 3}, /* 6400000 */ + {24000000, 4, 4}, /* 6000000 */ + {26000000, 4, 4}, /* 6500000 */ + {38400000, 6, 5} /* 6400000 */ +}; + +int cs42l74_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclkx_coeffs); i++) { + if (cs42l73_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +int cs42l74_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs42l73_mclk_coeffs); i++) { + if (cs42l73_mclk_coeffs[i].mclk == mclk && + cs42l73_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs42l73_set_mclk(struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk = 0; + u8 dmmcc = 0; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs42l74_get_mclkx_coeff(priv->sysclk); + + if (mclkx_coeff < 0) + return -EINVAL; + + mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / + cs42l73_mclkx_coeffs[mclkx_coeff].ratio; + + dev_dbg(codec->dev, "MCLK%u %u <-> internal MCLK %u\n", + priv->mclksel + 1, cs42l73_mclkx_coeffs[mclkx_coeff].mclkx, + mclk); + + dmmcc = + (priv->mclksel << 4) | (cs42l73_mclkx_coeffs[mclkx_coeff].mclkdiv << 1); + + cs42l73_write(codec, CS42L73_DMMCC, dmmcc); + + + priv->mclk = mclk; + + return 0; +} + +static int cs42l73_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + + if (clk_id != CS42L73_CLKID_MCLK1 && clk_id != CS42L73_CLKID_MCLK2) { + dev_err(codec->dev, "Invalid clk_id %u\n", clk_id); + return -EINVAL; + } + + if ((cs42l74_get_mclkx_coeff(freq) < 0)) { + dev_err(codec->dev, "Invalid sysclk %u\n", freq); + return -EINVAL; + } + + priv->sysclk = freq; + priv->mclksel = clk_id; + + return cs42l73_set_mclk(dai); +} + +/* + cs42l73_set_dai_clkdiv() + Setup MCLKx, MMCC dividers. + The dividers are selected from the MCLKX and the + sample rate. +*/ +static int cs42l73_set_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + int id = codec_dai->id; + u8 reg; + + switch (div_id) { + case CS42L73_MCLKXDIV: + /* MCLKDIV */ + reg = cs42l73_read(codec, CS42L73_DMMCC) & 0xf1; + cs42l73_write(codec, CS42L73_DMMCC, + reg | ((div & 0x07) << 1)); + break; + case CS42L73_MMCCDIV: + /* xSP MMCC */ + reg = cs42l73_read(codec, CS42L73_MMCC(id)) & 0xc0; + cs42l73_write(codec, CS42L73_MMCC(id), + reg | (div & 0x3f)); + break; + default: + return -EINVAL; + } + return 0; +} + +static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = codec_dai->id; + int inv, format; + u8 spc, mmcc; + + spc = cs42l73_read(codec, CS42L73_SPC(id)); + mmcc = cs42l73_read(codec, CS42L73_MMCC(id)); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mmcc |= MS_MASTER; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + mmcc &= ~MS_MASTER; + break; + + default: + return -EINVAL; + } + + format = (fmt & SND_SOC_DAIFMT_FORMAT_MASK); + inv = (fmt & SND_SOC_DAIFMT_INV_MASK); + + /* interface format */ + switch (format) { + case SND_SOC_DAIFMT_I2S: + spc &= ~xSPDIF_PCM; + break; + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + if (mmcc & MS_MASTER) { + dev_err(codec->dev, + "PCM format is supported only in slave mode\n"); + return -EINVAL; + } + if (id == CS42L73_ASP) { + dev_err(codec->dev, + "PCM format is not supported on ASP port\n"); + return -EINVAL; + } + spc |= xSPDIF_PCM; + break; + default: + return -EINVAL; + } + + if (spc & xSPDIF_PCM) { + spc &= (31 << 3); /* Clear PCM mode, set MSB->LSB */ + if (format == SND_SOC_DAIFMT_DSP_B + && inv == SND_SOC_DAIFMT_IB_IF) + spc |= (xPCM_MODE0 << 4); + else + + if (format == SND_SOC_DAIFMT_DSP_B + && inv == SND_SOC_DAIFMT_IB_NF) + spc |= (xPCM_MODE1 << 4); + else + + if (format == SND_SOC_DAIFMT_DSP_A + && inv == SND_SOC_DAIFMT_IB_IF) + spc |= (xPCM_MODE1 << 4); + else + return -EINVAL; + } + + priv->config[id].spc = spc; + priv->config[id].mmcc = mmcc; + + return 0; +} + +/* Sample rate converters */ +static u32 cs42l73_asrc_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static unsigned int cs42l73_get_xspfs_coeff(u32 rate) +{ + int i; + for (i = 0; i < ARRAY_SIZE(cs42l73_asrc_rates); i++) { + if (cs42l73_asrc_rates[i] == rate) + return (i + 1); + } + return 0; /* 0 = Don't know */ +} + +static void cs42l73_update_asrc(struct snd_soc_codec *codec, int id, int srate) +{ + u8 spfs = 0; + u8 reg; + + if (srate > 0) + spfs = cs42l73_get_xspfs_coeff(srate); + + switch (id) { + case CS42L73_XSP: + reg = cs42l73_read(codec, CS42L73_VXSPFS); + reg &= ~0x0f; + cs42l73_write(codec, CS42L73_VXSPFS, + reg | spfs ); + break; + case CS42L73_ASP: + reg = cs42l73_read(codec, CS42L73_ASPC); + reg &= ~0x3c; + cs42l73_write(codec, CS42L73_ASPC, + reg | (spfs << 2)); + break; + case CS42L73_VSP: + reg = cs42l73_read(codec, CS42L73_VXSPFS); + reg &= ~0xf0; + cs42l73_write(codec, CS42L73_VXSPFS, + reg | (spfs << 4) ); + break; + default: + break; + } +} + +static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + int id = dai->id; + int mclk_coeff; + int srate = params_rate(params); + + if (priv->config[id].mmcc & MS_MASTER) { + /* CS42L73 Master */ + /* MCLK -> srate */ + mclk_coeff = + cs42l74_get_mclk_coeff(priv->mclk, srate); + + if (mclk_coeff < 0) + return -EINVAL; + + dev_dbg(codec->dev, + "DAI[%d]: MCLK %u, srate %u, MMCC[5:0] = %x\n", + id, priv->mclk, srate, + cs42l73_mclk_coeffs[mclk_coeff].mmcc); + + priv->config[id].mmcc &= 0xC0; + priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= xMCK_SCLK_64FS; + + } else { + /* CS42L73 Slave */ + dev_dbg(codec->dev, "DAI[%d]: Slave\n", id); + priv->config[id].spc &= 0xFC; + priv->config[id].spc |= xMCK_SCLK_64FS; + } + /* Update ASRCs */ + priv->config[id].srate = srate; + cs42l73_update_asrc(codec, id, srate); + cs42l73_write(codec, CS42L73_SPC(id), priv->config[id].spc); + cs42l73_write(codec, CS42L73_MMCC(id), priv->config[id].mmcc); + + return 0; +} + +static int cs42l73_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 pwrctl1 = cs42l73_read(codec, CS42L73_PWRCTL1); + u8 dmmcc = cs42l73_read(codec, CS42L73_DMMCC); + + dev_dbg(codec->dev, + "%s: Level %d, PWRCTL1 0x%02x, DMMCC 0x%02x\n", + __FUNCTION__, level, pwrctl1, dmmcc); + + switch (level) { + case SND_SOC_BIAS_ON: + cs42l73_write(codec, CS42L73_DMMCC, dmmcc & ~ MCLKDIS); + cs42l73_write(codec, CS42L73_PWRCTL1, pwrctl1 & ~ PDN); + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + /* Powerdown all ports, inputs and outputs */ + cs42l73_write(codec, CS42L73_DMMCC, dmmcc & ~ MCLKDIS); + + cs42l73_write(codec, CS42L73_PWRCTL3, + PDN_THMS | PDN_SPKLO | PDN_EAR | + PDN_SPK | PDN_LO | PDN_HP); + + cs42l73_write(codec, CS42L73_PWRCTL2, + PDN_MIC2_BIAS | PDN_MIC1_BIAS | + PDN_VSP | PDN_ASP_SDOUT | PDN_ASP_SDIN | + PDN_XSP_SDOUT | PDN_XSP_SDIN); + + cs42l73_write(codec, CS42L73_PWRCTL1, + PDN_ADCB | PDN_ADCA | PDN_DMICB | + PDN_DMICA | PDN); + break; + + case SND_SOC_BIAS_OFF: + /* Powerdown all ports, inputs and outputs */ + cs42l73_write(codec, CS42L73_PWRCTL3, + PDN_THMS | PDN_SPKLO | PDN_EAR | + PDN_SPK | PDN_LO | PDN_HP); + + cs42l73_write(codec, CS42L73_PWRCTL2, + PDN_MIC2_BIAS | PDN_MIC1_BIAS | + PDN_VSP | PDN_ASP_SDOUT | PDN_ASP_SDIN | + PDN_XSP_SDOUT | PDN_XSP_SDIN); + + cs42l73_write(codec, CS42L73_PWRCTL1, + PDN_ADCB | PDN_ADCA | PDN_DMICB | + PDN_DMICA | PDN); + + cs42l73_write(codec, CS42L73_DMMCC, dmmcc | MCLKDIS); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +/* + cs42l73_set_tristate() + Tristate xSP SDOUT +*/ +static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + + u8 sp = cs42l73_read(codec, CS42L73_SPC(id)) & 0x7F; + + return cs42l73_write(codec, CS42L73_SPC(id), sp | (tristate << 7)); +} + +static void cs42l73_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + int id = dai->id; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + priv->config[id].srate = 0; + cs42l73_update_asrc(codec,id,0); +} + + +static struct snd_pcm_hw_constraint_list constraints_12_24 = { + .count = ARRAY_SIZE(cs42l73_asrc_rates), + .list = cs42l73_asrc_rates, +}; + +static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_12_24); + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + + +#define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops cs42l73_ops = { + .startup = cs42l73_pcm_startup, + .hw_params = cs42l73_pcm_hw_params, + .set_fmt = cs42l73_set_dai_fmt, + .set_sysclk = cs42l73_set_sysclk, + .set_clkdiv = cs42l73_set_clkdiv, + .set_tristate = cs42l73_set_tristate, + .shutdown = cs42l73_shutdown, +}; + +struct snd_soc_dai_driver cs42l73_dai[] = { + { + .name = "cs42l73-xsp", + .id = CS42L73_XSP, + .playback = { + .stream_name = "XSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + + .capture = { + .stream_name = "XSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-asp", + .id = CS42L73_ASP, + .playback = { + .stream_name = "ASP Playback", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + .capture = { + .stream_name = "ASP Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + }, + { + .name = "cs42l73-vsp", + .id = CS42L73_VSP, + .playback = { + .stream_name = "VSP Playback", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + .capture = { + .stream_name = "VSP Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS42L73_RATES, + .formats = CS42L73_FORMATS,}, + .ops = &cs42l73_ops, + .symmetric_rates = 1, + } +}; + +static int cs42l73_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int cs42l73_resume(struct snd_soc_codec *codec) +{ + int i; + u8 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = CS42L73_PWRCTL1; i < ARRAY_SIZE(cs42l73_reg); i++) { + cs42l73_write(codec, i, cache[i]); + } + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static int cs42l73_probe(struct snd_soc_codec *codec) +{ + int ret, i; + unsigned int devid = 0; + struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); + + codec->control_data = cs42l73->control_data; + codec->hw_write = (hw_write_t)i2c_master_send; + + cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* initialize codec */ + ret = cs42l73_read(codec, CS42L73_DEVID_AB); + devid = (ret & 0xFF) << 12; + + ret = cs42l73_read(codec, CS42L73_DEVID_CD); + devid |= (ret & 0xFF) << 4; + + ret = cs42l73_read(codec, CS42L73_DEVID_E); + devid |= (ret & 0xF0) >> 4; + + + if (devid != CS42L73_DEVID) { + dev_err(codec->dev, + "CS42L73 Device ID (%X). Expected %X\n", + devid, CS42L73_DEVID); + return ret; + } + + ret = cs42l73_read(codec, CS42L73_REVID); + if (ret < 0) { + dev_err(codec->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(codec->dev, + "Cirrus Logic CS42L73, Revision: %02X\n", ret & 0xFF); + + cs42l73->mclksel = CS42L73_CLKID_MCLK1; /* MCLK1 as master clk */ + cs42l73->mclk = 0; + + for (i = CS42L73_PWRCTL1; i < CS42L73_IS1; i++) + cs42l73_write(codec, i, cs42l73_reg[i]); + + snd_soc_add_controls(codec, cs42l73_snd_controls, + ARRAY_SIZE(cs42l73_snd_controls)); + + cs42l73_add_widgets(codec); + + return ret; +} + +static int cs42l73_remove(struct snd_soc_codec *codec) +{ + cs42l73_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +struct snd_soc_codec_driver soc_codec_dev_cs42l73 = { + .probe = cs42l73_probe, + .remove = cs42l73_remove, + .suspend = cs42l73_suspend, + .resume = cs42l73_resume, + .write = cs42l73_write, + .read = cs42l73_read, + .set_bias_level = cs42l73_set_bias_level, + .reg_cache_size = CS42L73_CACHEREGNUM, + .reg_cache_default = cs42l73_reg, +}; + +static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs42l73_private *cs42l73; + int ret; + + cs42l73 = kzalloc(sizeof(struct cs42l73_private), GFP_KERNEL); + if (!cs42l73) { + dev_err(&i2c_client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + i2c_set_clientdata(i2c_client, cs42l73); + cs42l73->control_data = i2c_client; + cs42l73->control_type = SND_SOC_I2C; + + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs42l73, cs42l73_dai, ARRAY_SIZE(cs42l73_dai)); + if (ret < 0) + kfree(cs42l73); + return ret; +} + +static __devexit int cs42l73_i2c_remove(struct i2c_client *client) +{ + struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + kfree(cs42l73); + + return 0; +} + +static const struct i2c_device_id cs42l73_id[] = { + {"cs42l73", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs42l73_id); + +static struct i2c_driver cs42l73_i2c_driver = { + .driver = { + .name = "cs42l73", + .owner = THIS_MODULE, + }, + .id_table = cs42l73_id, + .probe = cs42l73_i2c_probe, + .remove = __devexit_p(cs42l73_i2c_remove), + +}; + +static int __init cs42l73_modinit(void) +{ + int ret; + ret = i2c_add_driver(&cs42l73_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "%s: can't add i2c driver\n", __func__); + return ret; + } + return 0; +} + +module_init(cs42l73_modinit); + +static void __exit cs42l73_exit(void) +{ + i2c_del_driver(&cs42l73_i2c_driver); +} + +module_exit(cs42l73_exit); + +MODULE_DESCRIPTION("ASoC CS42L73 driver"); +MODULE_AUTHOR("Georgi Vlaev, Nucleus Systems Ltd, <office(a)nucleusys.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l73.h b/sound/soc/codecs/cs42l73.h new file mode 100644 index 0000000..c96843f --- /dev/null +++ b/sound/soc/codecs/cs42l73.h @@ -0,0 +1,225 @@ +/* + * ALSA SoC CS42L73 codec driver + * + * Copyright 2010 Cirrus Logic, Inc. + * + * Author: Georgi Vlaev <office(a)nucleusys.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#ifndef __CS42L73_H__ +#define __CS42L73_H__ + +/* I2C Registers */ +/* I2C Address: 1001010[R/W] - 10010100 = 0x94(Write); 10010101 = 0x95(Read) */ +#define CS42L73_CHIP_ID 0x4a +#define CS42L73_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS42L73_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS42L73_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS42L73_REVID 0x05 /* Revision ID [RO]. */ +#define CS42L73_PWRCTL1 0x06 /* Power Control 1. */ +#define CS42L73_PWRCTL2 0x07 /* Power Control 2. */ +#define CS42L73_PWRCTL3 0x08 /* Power Control 3, Thermal Overload Threshold. */ +#define CS42L73_CPFCHC 0x09 /* Charge Pump Freq. & Class H Control. */ +#define CS42L73_OLMBMSDC 0x0A /* Output Load, MIC Bias, & MIC2 Short Detect Config. */ +#define CS42L73_DMMCC 0x0B /* Digital MIC & Master Clock Control. */ +#define CS42L73_XSPC 0x0C /* Auxiliary Serial Port (XSP) Control. */ +#define CS42L73_XSPMMCC 0x0D /* XSP Master Mode Clocking Control. */ +#define CS42L73_ASPC 0x0E /* Audio Serial Port (ASP) Control. */ +#define CS42L73_ASPMMCC 0x0F /* ASP Master Mode Clocking Control. */ +#define CS42L73_VSPC 0x10 /* Voice Serial Port (VSP) Control. */ +#define CS42L73_VSPMMCC 0x11 /* VSP Master Mode Clocking Control. */ +#define CS42L73_VXSPFS 0x12 /* VSP & XSP Sample Rate. */ +#define CS42L73_MIOPC 0x13 /* Misc. Input & Output Path Control. */ +#define CS42L73_ADCIPC 0x14 /* ADC/IP Control. */ +#define CS42L73_MICAPREPGAAVOL 0x15 /* MIC 1 [A] PreAmp, PGAA Vol. */ +#define CS42L73_MICBPREPGABVOL 0x16 /* MIC 2 [B] PreAmp, PGAB Vol. */ +#define CS42L73_IPADVOL 0x17 /* Input Pat7h A Digital Volume. */ +#define CS42L73_IPBDVOL 0x18 /* Input Path B Digital Volume. */ +#define CS42L73_PBDC 0x19 /* Playback Digital Control. */ +#define CS42L73_HLADVOL 0x1A /* Headphone/Line A Out Digital Vol. */ +#define CS42L73_HLBDVOL 0x1B /* Headphone/Line B Out Digital Vol. */ +#define CS42L73_SPKDVOL 0x1C /* Speakerphone Out [A] Digital Vol. */ +#define CS42L73_ESLDVOL 0x1D /* Ear/Speakerphone Line Out [B] Digital Vol. */ +#define CS42L73_HPAAVOL 0x1E /* Headphone A Analog Volume. */ +#define CS42L73_HPBAVOL 0x1F /* Headphone B Analog Volume. */ +#define CS42L73_LOAAVOL 0x20 /* Line Out A Analog Volume. */ +#define CS42L73_LOBAVOL 0x21 /* Line Out B Analog Volume. */ +#define CS42L73_STRINV 0x22 /* Stereo Input Path Adv. Vol. */ +#define CS42L73_XSPINV 0x23 /* Auxiliary Serial Port Input Advisory Vol. */ +#define CS42L73_ASPINV 0x24 /* Audio Serial Port Input Advisory Vol. */ +#define CS42L73_VSPINV 0x25 /* Voice Serial Port Input Advisory Vol. */ +#define CS42L73_LIMARATEHL 0x26 /* Limiter Attack Rate Headphone/Line. */ +#define CS42L73_LIMRRATEHL 0x27 /* Limiter Ctl, Rel.Rate Headphone/Line. */ +#define CS42L73_LMAXHL 0x28 /* Limiter Thresholds Headphone/Line. */ +#define CS42L73_LIMARATESPK 0x29 /* Limiter Attack Rate Speakerphone [A]. */ +#define CS42L73_LIMRRATESPK 0x2A /* Limiter Ctl,Release Rate Speakerph. [A]. */ +#define CS42L73_LMAXSPK 0x2B /* Limiter Thresholds Speakerphone [A]. */ +#define CS42L73_LIMARATEESL 0x2C /* Limiter Attack Rate Ear/Speakerph.Line [B]. */ +#define CS42L73_LIMRRATEESL 0x2D /* Limiter Ctl,Release Rate Ear/Speakerphone Line [B]. */ +#define CS42L73_LMAXESL 0x2E /* Limiter Thresholds Ear/Speakerph. Line [B]. */ +#define CS42L73_ALCARATE 0x2F /* ALC Enable, Attack Rate AB. */ +#define CS42L73_ALCRRATE 0x30 /* ALC Release Rate AB. */ +#define CS42L73_ALCMINMAX 0x31 /* ALC Thresholds AB. */ +#define CS42L73_NGCAB 0x32 /* Noise Gate Ctl AB. */ +#define CS42L73_ALCNGMC 0x33 /* ALC & Noise Gate Misc Ctl. */ +#define CS42L73_MIXERCTL 0x34 /* Mixer Control. */ +#define CS42L73_HLAIPAA 0x35 /* HP/LO Left Mixer: Input Path Left Atten. */ +#define CS42L73_HLBIPBA 0x36 /* HP/LO Right Mixer: Input Path Rt. Atten. */ +#define CS42L73_HLAXSPAA 0x37 /* HP/LO Left Mixer: XSP Left Attenuation. */ +#define CS42L73_HLBXSPBA 0x38 /* HP/LO Right Mixer: XSP Rt. Attenuation. */ +#define CS42L73_HLAASPAA 0x39 /* HP/LO Left Mixer: ASP Left Attenuation. */ +#define CS42L73_HLBASPBA 0x3A /* HP/LO Right Mixer: ASP Rt. Attenuation. */ +#define CS42L73_HLAVSPMA 0x3B /* HP/LO Left Mixer: VSP Mono Atten. */ +#define CS42L73_HLBVSPMA 0x3C /* HP/LO Right Mixer: VSP Mono Atten. */ +#define CS42L73_XSPAIPAA 0x3D /* XSP Left Mixer: Input Path Left Attenuation. */ +#define CS42L73_XSPBIPBA 0x3E /* XSP Rt. Mixer: Input Path Right Attenuation. */ +#define CS42L73_XSPAXSPAA 0x3F /* XSP Left Mixer: XSP Left Attenuation. */ +#define CS42L73_XSPBXSPBA 0x40 /* XSP Rt. Mixer: XSP Right Attenuation. */ +#define CS42L73_XSPAASPAA 0x41 /* XSP Left Mixer: ASP Left Attenuation. */ +#define CS42L73_XSPAASPBA 0x42 /* XSP Rt. Mixer: ASP Right Attenuation. */ +#define CS42L73_XSPAVSPMA 0x43 /* XSP Left Mixer: VSP Mono Attenuation. */ +#define CS42L73_XSPBVSPMA 0x44 /* XSP Rt. Mixer: VSP Mono Attenuation. */ +#define CS42L73_ASPAIPAA 0x45 /* ASP Left Mixer: Input Path Left Attenuation. */ +#define CS42L73_ASPBIPBA 0x46 /* ASP Rt. Mixer: Input Path Right Attenuation. */ +#define CS42L73_ASPAXSPAA 0x47 /* ASP Left Mixer: XSP Left Attenuation. */ +#define CS42L73_ASPBXSPBA 0x48 /* ASP Rt. Mixer: XSP Right Attenuation. */ +#define CS42L73_ASPAASPAA 0x49 /* ASP Left Mixer: ASP Left Attenuation. */ +#define CS42L73_ASPBASPBA 0x4A /* ASP Rt. Mixer: ASP Right Attenuation. */ +#define CS42L73_ASPAVSPMA 0x4B /* ASP Left Mixer: VSP Mono Attenuation. */ +#define CS42L73_ASPBVSPMA 0x4C /* ASP Rt. Mixer: VSP Mono Attenuation. */ +#define CS42L73_VSPAIPAA 0x4D /* VSP Left Mixer: Input Path Left Attenuation. */ +#define CS42L73_VSPBIPBA 0x4E /* VSP Rt. Mixer: Input Path Right Attenuation. */ +#define CS42L73_VSPAXSPAA 0x4F /* VSP Left Mixer: XSP Left Attenuation. */ +#define CS42L73_VSPBXSPBA 0x50 /* VSP Rt. Mixer: XSP Right Attenuation. */ +#define CS42L73_VSPAASPAA 0x51 /* VSP Left Mixer: ASP Left Attenuation. */ +#define CS42L73_VSPBASPBA 0x52 /* VSP Rt. Mixer: ASP Right Attenuation. */ +#define CS42L73_VSPAVSPMA 0x53 /* VSP Left Mixer: VSP Mono Attenuation.*/ +#define CS42L73_VSPBVSPMA 0x54 /* VSP Rt. Mixer: VSP Mono Attenuation. */ +#define CS42L73_MMIXCTL 0x55 /* Mono Mixer Controls. */ +#define CS42L73_SPKMIPMA 0x56 /* SPK Mono Mixer: In. Path Mono Atten. */ +#define CS42L73_SPKMXSPA 0x57 /* SPK Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_SPKMASPA 0x58 /* SPK Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_SPKMVSPMA 0x59 /* SPK Mono Mixer: VSP Mono Atten. */ +#define CS42L73_ESLMIPMA 0x5A /* Ear/SpLO Mono Mixer: In. Path Mono Atten. */ +#define CS42L73_ESLMXSPA 0x5B /* Ear/SpLO Mono Mixer: XSP Mono/L/R Att. */ +#define CS42L73_ESLMASPA 0x5C /* Ear/SpLO Mono Mixer: ASP Mono/L/R Att. */ +#define CS42L73_ESLMVSPMA 0x5D /* Ear/SpLO Mono Mixer: VSP Mono Atten. */ +#define CS42L73_IM1 0x5E /* Interrupt Mask 1. */ +#define CS42L73_IM2 0x5F /* Interrupt Mask 2. */ +#define CS42L73_IS1 0x60 /* Interrupt Status 1 [RO]. */ +#define CS42L73_IS2 0x61 /* Interrupt Status 2 [RO]. */ + +#define CS42L73_CACHEREGNUM (CS42L73_IS2 + 1) + +/* Bitfield Definitions */ + +/* CS42L73_PWRCTL1 */ +#define PDN_ADCB (1 << 7) +#define PDN_DMICB (1 << 6) +#define PDN_ADCA (1 << 5) +#define PDN_DMICA (1 << 4) +#define PDN_LDO (1 << 2) +#define DISCHG_FILT (1 << 1) +#define PDN (1 << 0) + +/* CS42L73_PWRCTL2 */ +#define PDN_MIC2_BIAS (1 << 7) +#define PDN_MIC1_BIAS (1 << 6) +#define PDN_VSP (1 << 4) +#define PDN_ASP_SDOUT (1 << 3) +#define PDN_ASP_SDIN (1 << 2) +#define PDN_XSP_SDOUT (1 << 1) +#define PDN_XSP_SDIN (1 << 0) + +/* CS42L73_PWRCTL3 */ +#define PDN_THMS (1 << 5) +#define PDN_SPKLO (1 << 4) +#define PDN_EAR (1 << 3) +#define PDN_SPK (1 << 2) +#define PDN_LO (1 << 1) +#define PDN_HP (1 << 0) + +/* Thermal Overload Detect. Requires interrupt ... */ +#define THMOVLD_150C 0 +#define THMOVLD_132C 1 +#define THMOVLD_115C 2 +#define THMOVLD_098C 3 + + +/* CS42L73_ASPC, CS42L73_XSPC, CS42L73_VSPC */ +#define xSP_3ST (1 << 7) +#define xSPDIF_I2S 0 +#define xSPDIF_PCM (1 << 6) +#define xPCM_MODE0 0 +#define xPCM_MODE1 1 +#define xPCM_MODE2 2 +#define xPCM_BO_MSBLSB 0 +#define xPCM_BO_LSBMSB 1 +#define xMCK_SCLK_64FS 0 +#define xMCK_SCLK_MCLK 2 +#define xMCK_SCLK_PREMCLK 3 + +/* CS42L73_xSPMMCC */ +#define MS_MASTER (1 << 7) + + +/* CS42L73_DMMCC */ +#define MCLKDIS (1 << 0) +#define MCLKSEL_MCLK2 (1 << 4) +#define MCLKSEL_MCLK1 (0 << 4) + +/* CS42L73 MCLK derived from MCLK1 or MCLK2 */ +#define CS42L73_CLKID_MCLK1 0 +#define CS42L73_CLKID_MCLK2 1 + +#define CS42L73_MCLKXDIV 0 +#define CS42L73_MMCCDIV 1 + +#define CS42L73_XSP 0 +#define CS42L73_ASP 1 +#define CS42L73_VSP 2 + +/* IS1, IM1 */ +#define MIC2_SDET (1 << 6) +#define THMOVLD (1 << 4) +#define DIGMIXOVFL (1 << 3) +#define IPBOVFL (1 << 1) +#define IPAOVFL (1 << 0) + +/* Analog Softramp */ +#define ANLGOSFT (1 << 0) + +/* HP A/B Mute */ +#define HPMUTE (1 << 7) +/* LO A/B Mute */ +#define LOMUTE (1 << 7) +/* SPK Digital Mute */ +#define SPKDMUTE (1 << 2) + +/* Misc defines for codec */ +#define CS42L73_RESET_GPIO 143 + +#define CS42L73_DEVID 0x00042A73 +#define CS42L73_MCLKX_MIN 5644800 +#define CS42L73_MCLKX_MAX 38400000 + +#define CS42L73_SPC(id) (CS42L73_XSPC + (id << 1)) +#define CS42L73_MMCC(id) (CS42L73_XSPMMCC + (id << 1)) +#define CS42L73_SPFS(id) ((id == CS42L73_ASP) ? CS42L73_ASPC : CS42L73_VXSPFS) + +#endif /* __CS42L73_H__ */ -- 1.7.5.4

3 2

Re: [alsa-devel] [PATCH 1/2] ASoC: wm8995: Return -EINVAL if device ID mismatch
by Mark Brown 16 Sep '11

16 Sep '11

On Fri, Sep 16, 2011 at 10:46:33AM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied both, thanks.

1 0

[alsa-devel] [PATCH v2]ASoC:blackfin:add ad73311 card driver name
by Scott Jiang 16 Sep '11

16 Sep '11

Add driver_name in snd_soc_card, asoc will copy this name to driver field of snd_card. ALSA lib need this info to load the proper card config. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad73311.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 732a247..975c6e8 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -192,6 +192,7 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { static struct snd_soc_card bf5xx_ad73311 = { .name = "bfin-ad73311", + .driver_name = "AD73311", .probe = bf5xx_probe, .dai_link = &bf5xx_ad73311_dai[CONFIG_SND_BF5XX_SPORT_NUM], .num_links = 1, -- 1.7.0.4

3 6

Re: [alsa-devel] [PATCH 1/2] ASoC: wm8995: Return -EINVAL if device ID mismatch
by Liam Girdwood 16 Sep '11

16 Sep '11

On Fri, 2011-09-16 at 10:46 +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/codecs/wm8995.c | 1 + > 1 files changed, 1 insertions(+), 0 deletions(-) > > diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c > index 5ad873f..05f7795 100644 > --- a/sound/soc/codecs/wm8995.c > +++ b/sound/soc/codecs/wm8995.c > @@ -1642,6 +1642,7 @@ static int wm8995_probe(struct snd_soc_codec *codec) > > if (ret != 0x8995) { > dev_err(codec->dev, "Invalid device ID: %#x\n", ret); > + ret = -EINVAL; > goto err_reg_enable; > } > Both Acked-by: Liam Girdwood <lrg(a)ti.com>

1 0

Re: [alsa-devel] [PATCH] ASoC: tpa6130a2: Remove obsolete cleanup for clientdata
by Mark Brown 16 Sep '11

16 Sep '11

On Fri, Sep 09, 2011 at 04:47:00PM +0800, Axel Lin wrote: > The i2c core will clear the clientdata pointer automatically, > we don't have to set the `data' field to NULL in remove() or > if probe() failed anymore. Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: davinci-pcm: trivial: replace link with actual chan/link
by Ben Gardiner 16 Sep '11

16 Sep '11

The ambiguously named variable 'link' is used as a temporary throughout davinci-pcm -- its presence makes grepping (and groking) the code difficult. Replace link with the value of link in almost all sites. The exception is a couple places where the last-assigned link/chan needs to be returned by a function -- in these cases, rename to last_link. Signed-off-by: Ben Gardiner <bengardiner(a)nanometrics.ca> --- sound/soc/davinci/davinci-pcm.c | 123 +++++++++++++++++++-------------------- 1 files changed, 59 insertions(+), 64 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index a49e667..d5fe08c 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -180,7 +180,6 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - int link = prtd->asp_link[0]; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; @@ -198,7 +197,8 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) fifo_level = prtd->params->fifo_level; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", link, dma_pos, period_size); + "dma_ptr = %x period_size=%x\n", prtd->asp_link[0], dma_pos, + period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -222,17 +222,19 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) } acnt = prtd->params->acnt; - edma_set_src(link, src, INCR, W8BIT); - edma_set_dest(link, dst, INCR, W8BIT); + edma_set_src(prtd->asp_link[0], src, INCR, W8BIT); + edma_set_dest(prtd->asp_link[0], dst, INCR, W8BIT); - edma_set_src_index(link, src_bidx, src_cidx); - edma_set_dest_index(link, dst_bidx, dst_cidx); + edma_set_src_index(prtd->asp_link[0], src_bidx, src_cidx); + edma_set_dest_index(prtd->asp_link[0], dst_bidx, dst_cidx); if (!fifo_level) - edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, count, 1, 0, + ASYNC); else - edma_set_transfer_params(link, acnt, fifo_level, count, - fifo_level, ABSYNC); + edma_set_transfer_params(prtd->asp_link[0], acnt, fifo_level, + count, fifo_level, + ABSYNC); } static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) @@ -305,7 +307,6 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) unsigned int acnt = params->acnt; /* divide by 2 for ping/pong */ unsigned int ping_size = snd_pcm_lib_period_bytes(substream) >> 1; - int link = prtd->asp_link[1]; unsigned int fifo_level = prtd->params->fifo_level; unsigned int count; if ((data_type == 0) || (data_type > 4)) { @@ -316,28 +317,26 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) dma_addr_t asp_src_pong = iram_dma->addr + ping_size; ram_src_cidx = ping_size; ram_dst_cidx = -ping_size; - edma_set_src(link, asp_src_pong, INCR, W8BIT); + edma_set_src(prtd->asp_link[1], asp_src_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_src_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_src_index(link, data_type, data_type * fifo_level); + edma_set_src_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_src_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_src(link, runtime->dma_addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } else { dma_addr_t asp_dst_pong = iram_dma->addr + ping_size; ram_src_cidx = -ping_size; ram_dst_cidx = ping_size; - edma_set_dest(link, asp_dst_pong, INCR, W8BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_pong, INCR, W8BIT); - link = prtd->asp_link[0]; - edma_set_dest_index(link, data_type, data_type * fifo_level); - link = prtd->asp_link[1]; - edma_set_dest_index(link, data_type, data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[0], data_type, + data_type * fifo_level); + edma_set_dest_index(prtd->asp_link[1], data_type, + data_type * fifo_level); - link = prtd->ram_link; - edma_set_dest(link, runtime->dma_addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, runtime->dma_addr, INCR, W32BIT); } if (!fifo_level) { @@ -354,10 +353,9 @@ static int ping_pong_dma_setup(struct snd_pcm_substream *substream) count, fifo_level, ABSYNC); } - link = prtd->ram_link; - edma_set_src_index(link, ping_size, ram_src_cidx); - edma_set_dest_index(link, ping_size, ram_dst_cidx); - edma_set_transfer_params(link, ping_size, 2, + edma_set_src_index(prtd->ram_link, ping_size, ram_src_cidx); + edma_set_dest_index(prtd->ram_link, ping_size, ram_dst_cidx); + edma_set_transfer_params(prtd->ram_link, ping_size, 2, runtime->periods, 2, ASYNC); /* init master params */ @@ -406,32 +404,32 @@ static int request_ping_pong(struct snd_pcm_substream *substream, { dma_addr_t asp_src_ping; dma_addr_t asp_dst_ping; - int link; + int ret; struct davinci_pcm_dma_params *params = prtd->params; /* Request ram master channel */ - link = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, + ret = prtd->ram_channel = edma_alloc_channel(EDMA_CHANNEL_ANY, davinci_pcm_dma_irq, substream, prtd->params->ram_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request ram link channel */ - link = prtd->ram_link = edma_alloc_slot( + ret = prtd->ram_link = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; - link = prtd->asp_link[1] = edma_alloc_slot( + ret = prtd->asp_link[1] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit3; prtd->ram_link2 = -1; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - link = prtd->ram_link2 = edma_alloc_slot( + ret = prtd->ram_link2 = edma_alloc_slot( EDMA_CTLR(prtd->ram_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit4; } /* circle ping-pong buffers */ @@ -448,36 +446,33 @@ static int request_ping_pong(struct snd_pcm_substream *substream, asp_dst_ping = iram_dma->addr; } /* ping */ - link = prtd->asp_link[0]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[0], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[0], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[0], 0, 0); + edma_set_dest_index(prtd->asp_link[0], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f) | TCINTEN); prtd->asp_params.opt |= TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); /* pong */ - link = prtd->asp_link[1]; - edma_set_src(link, asp_src_ping, INCR, W16BIT); - edma_set_dest(link, asp_dst_ping, INCR, W16BIT); - edma_set_src_index(link, 0, 0); - edma_set_dest_index(link, 0, 0); + edma_set_src(prtd->asp_link[1], asp_src_ping, INCR, W16BIT); + edma_set_dest(prtd->asp_link[1], asp_dst_ping, INCR, W16BIT); + edma_set_src_index(prtd->asp_link[1], 0, 0); + edma_set_dest_index(prtd->asp_link[1], 0, 0); - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[1], &prtd->asp_params); prtd->asp_params.opt &= ~(TCCMODE | EDMA_TCC(0x3f)); /* interrupt after every pong completion */ prtd->asp_params.opt |= TCINTEN | TCCHEN | EDMA_TCC(prtd->ram_channel & 0x3f); - edma_write_slot(link, &prtd->asp_params); + edma_write_slot(prtd->asp_link[1], &prtd->asp_params); /* ram */ - link = prtd->ram_link; - edma_set_src(link, iram_dma->addr, INCR, W32BIT); - edma_set_dest(link, iram_dma->addr, INCR, W32BIT); + edma_set_src(prtd->ram_link, iram_dma->addr, INCR, W32BIT); + edma_set_dest(prtd->ram_link, iram_dma->addr, INCR, W32BIT); pr_debug("%s: audio dma channels/slots in use for ram:%u %u %u," "for asp:%u %u %u\n", __func__, prtd->ram_channel, prtd->ram_link, prtd->ram_link2, @@ -494,7 +489,7 @@ exit2: edma_free_channel(prtd->ram_channel); prtd->ram_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) @@ -502,22 +497,22 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) struct snd_dma_buffer *iram_dma; struct davinci_runtime_data *prtd = substream->runtime->private_data; struct davinci_pcm_dma_params *params = prtd->params; - int link; + int ret; if (!params) return -ENODEV; /* Request asp master DMA channel */ - link = prtd->asp_channel = edma_alloc_channel(params->channel, + ret = prtd->asp_channel = edma_alloc_channel(params->channel, davinci_pcm_dma_irq, substream, prtd->params->asp_chan_q); - if (link < 0) + if (ret < 0) goto exit1; /* Request asp link channels */ - link = prtd->asp_link[0] = edma_alloc_slot( + ret = prtd->asp_link[0] = edma_alloc_slot( EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); - if (link < 0) + if (ret < 0) goto exit2; iram_dma = (struct snd_dma_buffer *)substream->dma_buffer.private_data; @@ -537,17 +532,17 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(link, &prtd->asp_params); + edma_read_slot(prtd->asp_link[0], &prtd->asp_params); prtd->asp_params.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); - prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(link) << 5; - edma_write_slot(link, &prtd->asp_params); + prtd->asp_params.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; + edma_write_slot(prtd->asp_link[0], &prtd->asp_params); return 0; exit2: edma_free_channel(prtd->asp_channel); prtd->asp_channel = -1; exit1: - return link; + return ret; } static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -- 1.7.4.1

3 3

[alsa-devel] [PATCH] pulse: Install a PulseAudio config snippet into alsa.conf.d
by gmane＠colin.guthr.ie 16 Sep '11

16 Sep '11

From: Colin Guthrie <colin(a)mageia.org> --- configure.in | 19 ++++++++++++++++++- pulse/Makefile.am | 2 ++ pulse/pulseaudio-default.conf.example | 13 +++++++++++++ pulse/pulseaudio.conf | 13 +++++++++++++ 4 files changed, 46 insertions(+), 1 deletions(-) create mode 100644 pulse/pulseaudio-default.conf.example create mode 100644 pulse/pulseaudio.conf diff --git a/configure.in b/configure.in index ccf59ba..3839dd5 100644 --- a/configure.in +++ b/configure.in @@ -131,10 +131,10 @@ fi AM_CONDITIONAL(HAVE_PPH, test "$PPH" = "builtin" -o "$PPH" = "lib") AM_CONDITIONAL(USE_LIBSPEEX, test "$PPH" = "lib") -dnl ALSA plugin directory test "x$prefix" = xNONE && prefix=$ac_default_prefix test "x$exec_prefix" = xNONE && exec_prefix=$prefix +dnl ALSA plugin directory AC_ARG_WITH(plugindir, AS_HELP_STRING([--with-plugindir=dir], [path where ALSA plugin files are stored]), @@ -151,6 +151,23 @@ AC_DEFINE_UNQUOTED(ALSA_PLUGIN_DIR, "$plugindir", [directory containing ALSA add ALSA_PLUGIN_DIR="$plugindir" AC_SUBST(ALSA_PLUGIN_DIR) +dnl ALSA data directory +AC_ARG_WITH(alsadatadir, + AS_HELP_STRING([--with-alsadatadir=dir], + [path where ALSA data files are stored]), + alsadatadir="$withval", alsadatadir="") +if test -z "$alsadatadir"; then + eval dir="$datadir" + case "$dir" in + /*) ;; + *) dir="$dir" + esac + alsadatadir="$dir/alsa" +fi +AC_DEFINE_UNQUOTED(ALSA_DATA_DIR, "$alsadatadir", [directory containing ALSA data files]) +ALSA_DATA_DIR="$alsadatadir" +AC_SUBST(ALSA_DATA_DIR) + SAVE_PLUGINS_VERSION AC_OUTPUT([ diff --git a/pulse/Makefile.am b/pulse/Makefile.am index e89e8ee..cca49ac 100644 --- a/pulse/Makefile.am +++ b/pulse/Makefile.am @@ -1,10 +1,12 @@ asound_module_pcm_LTLIBRARIES = libasound_module_pcm_pulse.la asound_module_ctl_LTLIBRARIES = libasound_module_ctl_pulse.la asound_module_conf_LTLIBRARIES = libasound_module_conf_pulse.la +asound_module_data_DATA = pulseaudio.conf pulseaudio-default.conf.example asound_module_pcmdir = @ALSA_PLUGIN_DIR@ asound_module_ctldir = @ALSA_PLUGIN_DIR@ asound_module_confdir = @ALSA_PLUGIN_DIR@ +asound_module_datadir = @ALSA_DATA_DIR@/alsa.conf.d AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ $(PTHREAD_CFLAGS) $(pulseaudio_CFLAGS) -D_GNU_SOURCE AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED) diff --git a/pulse/pulseaudio-default.conf.example b/pulse/pulseaudio-default.conf.example new file mode 100644 index 0000000..4f58858 --- /dev/null +++ b/pulse/pulseaudio-default.conf.example @@ -0,0 +1,13 @@ +# Default to PulseAudio + +pcm.!default { + type pulse + hint { + show on + description "Default ALSA Output (currently PulseAudio Sound Server)" + } +} + +ctl.!default { + type pulse +} diff --git a/pulse/pulseaudio.conf b/pulse/pulseaudio.conf new file mode 100644 index 0000000..dd85dab --- /dev/null +++ b/pulse/pulseaudio.conf @@ -0,0 +1,13 @@ +# Add a specific named PulseAudio pcm and ctl (typically useful for testing) + +pcm.pulse { + type pulse + hint { + show on + description "PulseAudio Sound Server" + } +} + +ctl.pulse { + type pulse +} -- 1.7.6

3 2

[PATCH] ALSA: HDA: Fix Realtek automute for Headphone to Line Out
by David Henningsson 16 Sep '11

16 Sep '11

If the machine has "Headphone out" and "Line Out", but no "Speaker out", the Realtek autoparser currently creates an Automute kcontrol, but it isn't working. This patch enables automuting of "Line out" when headphones are plugged, provided that the automute kcontrol is set to "enabled". BugLink: http://bugs.launchpad.net/bugs/851697 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 92 ++++++++++++++++++++++++----------------- 1 files changed, 54 insertions(+), 38 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 70ba45e..8dff8d2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -167,10 +167,12 @@ struct alc_spec { unsigned int master_mute:1; unsigned int auto_mic:1; unsigned int auto_mic_valid_imux:1; /* valid imux for auto-mic */ - unsigned int automute:1; /* HP automute enabled */ + unsigned int automute:1; /* Speaker automute enabled */ + unsigned int detect_hp:1; /* Headphones detection enabled */ unsigned int detect_line:1; /* Line-out detection enabled */ unsigned int automute_lines:1; /* automute line-out as well */ - unsigned int automute_hp_lo:1; /* both HP and LO available */ + unsigned int automute_speaker_possible:1; /* there are speakers and either LO or HP */ + unsigned int automute_lo_possible:1; /* there are line outs and HP */ /* other flags */ unsigned int no_analog :1; /* digital I/O only */ @@ -556,7 +558,7 @@ static void update_speakers(struct hda_codec *codec) if (spec->autocfg.line_out_pins[0] == spec->autocfg.hp_pins[0] || spec->autocfg.line_out_pins[0] == spec->autocfg.speaker_pins[0]) return; - if (!spec->automute_lines || !spec->automute) + if (!spec->automute_lines) on = 0; else on = spec->jack_present; @@ -582,7 +584,7 @@ static void alc_hp_automute(struct hda_codec *codec) spec->jack_present = detect_jacks(codec, ARRAY_SIZE(spec->autocfg.hp_pins), spec->autocfg.hp_pins); - if (!spec->automute) + if (!spec->detect_hp || (!spec->automute && !spec->automute_lines)) return; call_update_speakers(codec); } @@ -795,7 +797,7 @@ static int alc_automute_mode_info(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; - if (spec->automute_hp_lo) { + if (spec->automute_speaker_possible && spec->automute_lo_possible) { uinfo->value.enumerated.items = 3; texts = texts3; } else { @@ -814,13 +816,12 @@ static int alc_automute_mode_get(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - unsigned int val; - if (!spec->automute) - val = 0; - else if (!spec->automute_lines) - val = 1; - else - val = 2; + unsigned int val = 0; + if (spec->automute) + val++; + if (spec->automute_lines) + val++; + ucontrol->value.enumerated.item[0] = val; return 0; } @@ -833,18 +834,29 @@ static int alc_automute_mode_put(struct snd_kcontrol *kcontrol, switch (ucontrol->value.enumerated.item[0]) { case 0: - if (!spec->automute) + if (!spec->automute && !spec->automute_lines) return 0; spec->automute = 0; + spec->automute_lines = 0; break; case 1: - if (spec->automute && !spec->automute_lines) - return 0; - spec->automute = 1; - spec->automute_lines = 0; + if (spec->automute_speaker_possible) { + if (!spec->automute_lines && spec->automute) + return 0; + spec->automute = 1; + spec->automute_lines = 0; + } + else if (spec->automute_lo_possible) { + if (spec->automute_lines) + return 0; + spec->automute_lines = 1; + } + else + return -EINVAL; + break; case 2: - if (!spec->automute_hp_lo) + if (!spec->automute_lo_possible || !spec->automute_speaker_possible) return -EINVAL; if (spec->automute && spec->automute_lines) return 0; @@ -906,8 +918,6 @@ static void alc_init_auto_hp(struct hda_codec *codec) present++; if (present < 2) /* need two different output types */ return; - if (present == 3) - spec->automute_hp_lo = 1; /* both HP and LO automute */ if (!cfg->speaker_pins[0] && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { @@ -923,6 +933,8 @@ static void alc_init_auto_hp(struct hda_codec *codec) cfg->hp_outs = cfg->line_outs; } + spec->automute_mode = ALC_AUTOMUTE_PIN; + for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; if (!is_jack_detectable(codec, nid)) @@ -932,28 +944,32 @@ static void alc_init_auto_hp(struct hda_codec *codec) snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT); - spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_PIN; - } - if (spec->automute && cfg->line_out_pins[0] && - cfg->speaker_pins[0] && - cfg->line_out_pins[0] != cfg->hp_pins[0] && - cfg->line_out_pins[0] != cfg->speaker_pins[0]) { - for (i = 0; i < cfg->line_outs; i++) { - hda_nid_t nid = cfg->line_out_pins[i]; - if (!is_jack_detectable(codec, nid)) - continue; - snd_printdd("realtek: Enable Line-Out auto-muting " - "on NID 0x%x\n", nid); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC_FRONT_EVENT); - spec->detect_line = 1; + spec->detect_hp = 1; + } + + if (cfg->line_out_type == AUTO_PIN_LINE_OUT && cfg->line_outs) { + if (cfg->speaker_outs) + for (i = 0; i < cfg->line_outs; i++) { + hda_nid_t nid = cfg->line_out_pins[i]; + if (!is_jack_detectable(codec, nid)) + continue; + snd_printdd("realtek: Enable Line-Out " + "auto-muting on NID 0x%x\n", nid); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC_FRONT_EVENT); + spec->detect_line = 1; } - spec->automute_lines = spec->detect_line; + spec->automute_lo_possible = spec->detect_hp; } - if (spec->automute) { + spec->automute_speaker_possible = cfg->speaker_outs && + (spec->detect_hp || spec->detect_line); + + spec->automute_lines = spec->automute_lo_possible; + spec->automute = spec->automute_speaker_possible; + + if (spec->automute_speaker_possible || spec->automute_lo_possible) { /* create a control for automute mode */ alc_add_automute_mode_enum(codec); spec->unsol_event = alc_sku_unsol_event; -- 1.7.5.4 --------------050207030404040202010807--

1 0

[alsa-devel] [PATCH v3 1/1] ASoC: mxs-saif: add record function
by Dong Aisheng 16 Sep '11

16 Sep '11

1. add different clkmux mode handling SAIF can use two instances to implement full duplex (playback & recording) and record saif may work on EXTMASTER mode which is using other saif's BITCLK&LRCLK. The clkmux mode could be set in pdata->init() in mach-specific code. For generic saif driver, it only needs to know who is his master and the master id is also provided in mach-specific code. 2. support playback and capture simutaneously however the sample rates can not be different due to hw limitation. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Wolfram Sang <w.sang(a)pengutronix.de> --- Changes since v2: * remove mach-specific code(clkmux in DIGCTL) out of saif driver For supporting EXTMASTER mode, SAIF only nees to know who's its master and the master id is provided by mach layer according to different clkmux setting. So we need to add a pdata->get_master_id(); Changes since v1: * calc the delay based on the rate * change the saif.h #ifndef micro to match the directory --- include/sound/saif.h | 16 +++++ sound/soc/mxs/mxs-saif.c | 145 +++++++++++++++++++++++++++++++++++++++++----- sound/soc/mxs/mxs-saif.h | 4 + 3 files changed, 151 insertions(+), 14 deletions(-) diff --git a/include/sound/saif.h b/include/sound/saif.h new file mode 100644 index 0000000..d0e0de7 --- /dev/null +++ b/include/sound/saif.h @@ -0,0 +1,16 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __SOUND_SAIF_H__ +#define __SOUND_SAIF_H__ + +struct mxs_saif_platform_data { + int (*init) (void); + int (*get_master_id) (unsigned int saif_id); +}; +#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index af5734f..401944c 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -23,10 +23,12 @@ #include <linux/dma-mapping.h> #include <linux/clk.h> #include <linux/delay.h> +#include <linux/time.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/saif.h> #include <mach/dma.h> #include <asm/mach-types.h> #include <mach/hardware.h> @@ -36,6 +38,24 @@ static struct mxs_saif *mxs_saif[2]; +/* + * SAIF is a little different with other normal SOC DAIs on clock using. + * + * For MXS, two SAIF modules are instantiated on-chip. + * Each SAIF has a set of clock pins and can be operating in master + * mode simultaneously if they are connected to different off-chip codecs. + * Also, one of the two SAIFs can master or drive the clock pins while the + * other SAIF, in slave mode, receives clocking from the master SAIF. + * This also means that both SAIFs must operate at the same sample rate. + * + * We abstract this as each saif has a master, the master could be + * himself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is his master + * and operating his master to generate the proper clock rate for him. + * The master id is provided in mach-specific layer according to different + * clkmux setting. + */ + static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -52,6 +72,17 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, } /* + * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK + * is provided by other SAIF, we provide a interface here to get its master + * from its master_id. + * Note that the master could be himself. + */ +static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) +{ + return mxs_saif[saif->master_id]; +} + +/* * Set SAIF clock and MCLK */ static int mxs_saif_set_clk(struct mxs_saif *saif, @@ -60,8 +91,26 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, { u32 scr; int ret; + struct mxs_saif *master_saif; - scr = __raw_readl(saif->base + SAIF_CTRL); + dev_dbg(saif->dev, "mclk %d rate %d\n", mclk, rate); + + /* Set master saif to generate proper clock */ + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + dev_dbg(saif->dev, "master saif%d\n", master_saif->id); + + /* Checking if can playback and capture simutaneously */ + if (master_saif->ongoing && rate != master_saif->cur_rate) { + dev_err(saif->dev, + "can not change clock, master saif%d(rate %d) is ongoing\n", + master_saif->id, master_saif->cur_rate); + return -EINVAL; + } + + scr = __raw_readl(master_saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE; scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; @@ -75,27 +124,29 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * * If MCLK is not used, we just set saif clk to 512*fs. */ - if (saif->mclk_in_use) { + if (master_saif->mclk_in_use) { if (mclk % 32 == 0) { scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); } else if (mclk % 48 == 0) { scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 384 * rate); + ret = clk_set_rate(master_saif->clk, 384 * rate); } else { /* SAIF MCLK should be either 32x or 48x */ return -EINVAL; } } else { - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; } if (ret) return ret; - if (!saif->mclk_in_use) { - __raw_writel(scr, saif->base + SAIF_CTRL); + master_saif->cur_rate = rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -137,7 +188,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, return -EINVAL; } - __raw_writel(scr, saif->base + SAIF_CTRL); + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -183,6 +234,7 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, struct mxs_saif *saif = mxs_saif[saif_id]; u32 stat; int ret; + struct mxs_saif *master_saif; if (!saif) return -EINVAL; @@ -195,6 +247,12 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, __raw_writel(BM_SAIF_CTRL_CLKGATE, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + master_saif = mxs_saif_get_master(saif); + if (saif != master_saif) { + dev_err(saif->dev, "can not get mclk from a non-master saif\n"); + return -EINVAL; + } + stat = __raw_readl(saif->base + SAIF_STAT); if (stat & BM_SAIF_STAT_BUSY) { dev_err(saif->dev, "error: busy\n"); @@ -278,10 +336,17 @@ static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* * Note: We simply just support master mode since SAIF TX can only * work as master. + * Here the master is relative to codec side. + * Saif internally could be slave when working on EXTMASTER mode. + * We just hide this to machine driver. */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + if (saif->id == saif->master_id) + scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + else + scr |= BM_SAIF_CTRL_SLAVE_MODE; + __raw_writel(scr | scr0, saif->base + SAIF_CTRL); break; default: @@ -396,6 +461,12 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; + u32 delay; + + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -403,10 +474,20 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(saif->clk); - if (!saif->mclk_in_use) + clk_enable(master_saif->clk); + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + + /* + * If the saif's master is not himself, we also need to enable + * itself clk for its internal basic logic to work. + */ + if (saif != master_saif) { + clk_enable(saif->clk); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* @@ -422,20 +503,39 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, __raw_readl(saif->base + SAIF_DATA); } - dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n", + master_saif->ongoing = 1; + + dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", __raw_readl(saif->base + SAIF_CTRL), __raw_readl(saif->base + SAIF_STAT)); + dev_dbg(master_saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(master_saif->base + SAIF_CTRL), + __raw_readl(master_saif->base + SAIF_STAT)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dev_dbg(cpu_dai->dev, "stop\n"); - clk_disable(saif->clk); - if (!saif->mclk_in_use) + /* wait a while for the current sample to complete */ + delay = USEC_PER_SEC / master_saif->cur_rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + } + clk_disable(master_saif->clk); + + if (saif != master_saif) { __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + clk_disable(saif->clk); + } + + master_saif->ongoing = 0; break; default: @@ -519,16 +619,33 @@ static int mxs_saif_probe(struct platform_device *pdev) { struct resource *res; struct mxs_saif *saif; + struct mxs_saif_platform_data *pdata; int ret = 0; if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; + pdata = pdev->dev.platform_data; + if (pdata && pdata->init) { + ret = pdata->init(); + if (ret) + return ret; + } + saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; mxs_saif[pdev->id] = saif; + saif->id = pdev->id; + + saif->master_id = saif->id; + if (pdata && pdata->get_master_id) { + saif->master_id = pdata->get_master_id(saif->id); + if (saif->master_id < 0 || + saif->master_id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + } saif->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 0e2ff8c..12c91e4 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -118,6 +118,10 @@ struct mxs_saif { void __iomem *base; int irq; struct mxs_pcm_dma_params dma_param; + unsigned int id; + unsigned int master_id; + unsigned int cur_rate; + unsigned int ongoing; struct platform_device *soc_platform_pdev; u32 fifo_underrun; -- 1.7.0.4

6 7

[alsa-devel] snd_pcm_pause kills plaback handle on OMAP4
by john＠johnandjuliet.com 16 Sep '11

16 Sep '11

I'm trying to control the McBSP2 that is used for the modem in an android environment (BLAZE OMAP4) 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 .. 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. This implies some sort of bug in the omap soc driver? Any help on how to figure this out? Thnkx JBP

2 1

[alsa-devel] Allow alsa configuration to be loaded from a alsa.conf.d folder
by gmane＠colin.guthr.ie 16 Sep '11

16 Sep '11

Hi, This is a small patch for alsa-lib that supports the common ".d" folder configs. The idea here is to make it easier for packagers and distros to supply plugin configuration without having to patch alsa-lib to do so (or by managing a /etc/asound.conf file which may want to be tweaked by multiple files) If this patch is accepted, I'll modify alsa-plugins to inlcude a pulseaudio.conf file in /usr/share/alsa/alsa.conf.d/ which defines a pcm.pulse anc ctl.pulse. If distros want to default to PA, they should include their own file in there that defines pcm|ctl.!default appropriately. Similar configuration could be added for third party alsa plugins such as the DTS encoder plugin being developed by Alexander Patrikov which are shipped outside of alsa. I don't think there is anything too controversial in here, so hopefully it'll be accepted. Cheers Col

4 9

[alsa-devel] snd-usb: endpoint code rework
by Daniel Mack 16 Sep '11

16 Sep '11

With major parts of code shuffling out of the way, it might get clear where the actual rework of the endpoint code goes. I've attached my local diff (basically the same code base as some weeks ago) for an review of the rough idea. Thanks, Daniel

2 3

[alsa-devel] [PATCH] snd-usb-6fire: volume control for individual analog channels
by Torsten Schenk 16 Sep '11

16 Sep '11

Added a stereo volume control for each analog output channel pair. Master volume modified so that it acts as a software weight for the individual channel faders. Added mute switches for master and individual analog channels. Signed-off-by: Torsten Schenk <torsten.schenk(a)zoho.com> Cc: stable(a)kernel.org [3.0+] --- diff -Nur a/sound/usb/6fire/chip.c b/sound/usb/6fire/chip.c --- a/sound/usb/6fire/chip.c 2011-09-15 16:11:08.000000000 +0000 +++ b/sound/usb/6fire/chip.c 2011-09-15 16:11:46.000000000 +0000 @@ -5,7 +5,7 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 + * Version: 0.3.6 * Copyright: (C) Torsten Schenk * * This program is free software; you can redistribute it and/or modify @@ -29,7 +29,7 @@ #include <sound/initval.h> MODULE_AUTHOR("Torsten Schenk <torsten.schenk(a)zoho.com>"); -MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.0"); +MODULE_DESCRIPTION("TerraTec DMX 6Fire USB audio driver, version 0.3.6"); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{TerraTec, DMX 6Fire USB}}"); diff -Nur a/sound/usb/6fire/control.c b/sound/usb/6fire/control.c --- a/sound/usb/6fire/control.c 2011-09-02 13:48:10.000000000 +0000 +++ b/sound/usb/6fire/control.c 2011-09-02 13:50:35.000000000 +0000 @@ -5,9 +5,13 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 + * Version: 0.3.6 * Copyright: (C) Torsten Schenk * + * Thanks to: + * - Holger Ruckdeschel: he found out how to control individual analog + * channel volumes and introduced mute switches + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -59,7 +63,7 @@ { 0x22, 0x03, 0x00 }, { 0x20, 0x03, 0x08 }, { 0x22, 0x04, 0x00 }, { 0x20, 0x04, 0x08 }, { 0x22, 0x05, 0x01 }, { 0x20, 0x05, 0x08 }, { 0x22, 0x04, 0x01 }, { 0x12, 0x04, 0x00 }, { 0x12, 0x05, 0x00 }, - { 0x12, 0x0d, 0x78 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, + { 0x12, 0x0d, 0x38 }, { 0x12, 0x21, 0x82 }, { 0x12, 0x22, 0x80 }, { 0x12, 0x23, 0x00 }, { 0x12, 0x06, 0x02 }, { 0x12, 0x03, 0x00 }, { 0x12, 0x02, 0x00 }, { 0x22, 0x03, 0x01 }, { 0 } /* TERMINATING ENTRY */ @@ -74,15 +78,35 @@ DIGITAL_THRU_ONLY_SAMPLERATE = 3 }; -static void usb6fire_control_master_vol_update(struct control_runtime *rt) +static void usb6fire_control_output_vol_update(struct control_runtime *rt) +{ + struct comm_runtime *comm_rt = rt->chip->comm; + u8 value; + + if (comm_rt) { + int i; + for (i = 0; i < 6; i++) { + /* set volume */ + value = rt->output_vol[i] * rt->master_vol / 0x7f; + comm_rt->write8(comm_rt, 0x12, 0x0f + i, 0x7f - + log_volume_table[value]); + } + } +} + +static void usb6fire_control_output_switch_update(struct control_runtime *rt) { struct comm_runtime *comm_rt = rt->chip->comm; + int i; + u8 value = 0x00; + if (comm_rt) { - /* set volume */ - comm_rt->write8(comm_rt, 0x12, 0x0f, 0x7f - - log_volume_table[rt->master_vol]); - /* unmute */ - comm_rt->write8(comm_rt, 0x12, 0x0e, 0x00); + if (rt->master_switch) /* mute/unmute single outputs */ + for (i = 0; i < 6; i++) + value |= !rt->output_switch[i] << i; + else /* mute all outputs */ + value = 0x3f; + comm_rt->write8(comm_rt, 0x12, 0x0e, value); } } @@ -171,7 +195,17 @@ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = 127; + uinfo->value.integer.max = 0x7f; + return 0; +} + +static int usb6fire_control_output_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0x7f; return 0; } @@ -182,7 +216,12 @@ int changed = 0; if (rt->master_vol != ucontrol->value.integer.value[0]) { rt->master_vol = ucontrol->value.integer.value[0]; - usb6fire_control_master_vol_update(rt); + usb6fire_control_output_vol_update(rt); + changed = 1; + } + if (rt->master_vol != ucontrol->value.integer.value[0]) { + rt->master_vol = ucontrol->value.integer.value[0]; + usb6fire_control_output_vol_update(rt); changed = 1; } return changed; @@ -196,6 +235,127 @@ return 0; } +static int usb6fire_control_master_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + int changed = 0; + if (rt->master_switch != ucontrol->value.integer.value[0]) { + rt->master_switch = ucontrol->value.integer.value[0]; + usb6fire_control_output_switch_update(rt); + changed = 1; + } + return changed; +} + +static int usb6fire_control_master_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + ucontrol->value.integer.value[0] = rt->master_switch; + return 0; +} + +static int usb6fire_control_output_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + int changed = 0; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + if (rt->output_vol[ch0] != ucontrol->value.integer.value[0]) { + rt->output_vol[ch0] = ucontrol->value.integer.value[0]; + changed = 1; + } + if (rt->output_vol[ch1] != ucontrol->value.integer.value[1]) { + rt->output_vol[ch1] = ucontrol->value.integer.value[1]; + changed = 1; + } + + if (changed) + usb6fire_control_output_vol_update(rt); + + return changed; +} + +static int usb6fire_control_output_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + ucontrol->value.integer.value[0] = rt->output_vol[ch0]; + ucontrol->value.integer.value[1] = rt->output_vol[ch1]; + + return 0; +} + +static int usb6fire_control_output_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + int changed = 0; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + if (rt->output_switch[ch0] != ucontrol->value.integer.value[0]) { + rt->output_switch[ch0] = ucontrol->value.integer.value[0]; + changed = 1; + } + if (rt->output_switch[ch1] != ucontrol->value.integer.value[1]) { + rt->output_switch[ch1] = ucontrol->value.integer.value[1]; + changed = 1; + } + + if (changed) + usb6fire_control_output_switch_update(rt); + + return changed; +} + +static int usb6fire_control_output_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct control_runtime *rt = snd_kcontrol_chip(kcontrol); + unsigned long chs = kcontrol->private_value; + unsigned long ch0 = 2 * chs; + unsigned long ch1 = 2 * chs + 1; + + if (chs < 0 || chs > 2) { + snd_printk(KERN_ERR PREFIX + "Invalid channel in volume control."); + return 0; + } + + ucontrol->value.integer.value[0] = rt->output_switch[ch0]; + ucontrol->value.integer.value[1] = rt->output_switch[ch1]; + + return 0; +} + static int usb6fire_control_line_phono_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -299,6 +459,75 @@ }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = usb6fire_control_master_switch_get, + .put = usb6fire_control_master_switch_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 1/2 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 1/2 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_switch_get, + .put = usb6fire_control_output_switch_put, + .private_value = 0 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 3/4 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 1 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 3/4 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_switch_get, + .put = usb6fire_control_output_switch_put, + .private_value = 1 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 5/6 Playback Volume", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = usb6fire_control_output_vol_info, + .get = usb6fire_control_output_vol_get, + .put = usb6fire_control_output_vol_put, + .private_value = 2 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Output 5/6 Playback Switch", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_stereo_info, + .get = usb6fire_control_output_switch_get, + .put = usb6fire_control_output_switch_put, + .private_value = 2 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Line/Phono Capture Route", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -352,7 +581,8 @@ usb6fire_control_opt_coax_update(rt); usb6fire_control_line_phono_update(rt); - usb6fire_control_master_vol_update(rt); + usb6fire_control_output_vol_update(rt); + usb6fire_control_output_switch_update(rt); usb6fire_control_streaming_update(rt); i = 0; @@ -378,3 +608,4 @@ kfree(chip->control); chip->control = NULL; } + diff -Nur a/sound/usb/6fire/control.h b/sound/usb/6fire/control.h --- a/sound/usb/6fire/control.h 2011-09-02 13:48:10.000000000 +0000 +++ b/sound/usb/6fire/control.h 2011-09-02 13:50:30.000000000 +0000 @@ -3,9 +3,13 @@ * * Author: Torsten Schenk <torsten.schenk(a)zoho.com> * Created: Jan 01, 2011 - * Version: 0.3.0 + * Version: 0.3.6 * Copyright: (C) Torsten Schenk * + * Thanks to: + * - Holger Ruckdeschel: he found out how to control individual analog + * channel volumes and introduced mute switches + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -45,6 +49,9 @@ bool digital_thru_switch; bool usb_streaming; u8 master_vol; + bool master_switch; + u8 output_vol[6]; + bool output_switch[6]; }; int __devinit usb6fire_control_init(struct sfire_chip *chip);

3 3

[alsa-devel] Temporary locations of sound git and snapshot tarballs
by Takashi Iwai 16 Sep '11

16 Sep '11

Hi, as kernel.org downtime continues unexpectedly long, I've set up a tentative git repo in below as already mentioned in another post: git://github.com/tiwai/sound.git Also, the corresponding alsa-driver-build tree is found in git://github.com/tiwai/alsa-driver-build.git The snapshot tarballs are found in: ftp://ftp.suse.com/pub/people/tiwai/snapshot/alsa-driver-snapshot.tar.gz No daily snapshots are found there, and only *.gz is present. thanks, Takashi

1 0

[alsa-devel] [PATCH] ASoC: bf5xx-ad73311: Fix prototype for bf5xx_probe
by Axel Lin 16 Sep '11

16 Sep '11

Fix below build warning: sound/soc/blackfin/bf5xx-ad73311.c: warning: initialization from incompatible pointer type Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad73311.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c index 732a247..b94eb7e 100644 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ b/sound/soc/blackfin/bf5xx-ad73311.c @@ -128,7 +128,7 @@ static int snd_ad73311_configure(void) return 0; } -static int bf5xx_probe(struct platform_device *pdev) +static int bf5xx_probe(struct snd_soc_card *card) { int err; if (gpio_request(GPIO_SE, "AD73311_SE")) { -- 1.7.4.1

3 2

[alsa-devel] [PATCH v2] ASoC: samsung: Fix checking return value of clk_get
by Axel Lin 16 Sep '11

16 Sep '11

clk_get() returns a pointer to the struct clk or an ERR_PTR(). This patch also use PTR_ERR() for return value. Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Christian Pellegrin <chripell(a)evolware.org> Cc: Ben Dooks <ben(a)simtec.co.uk> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Jassi Brar <jassisinghbrar(a)gmail.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- v2: Also use PTR_ERR() for return value. sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx_uda134x.c | 8 ++++---- 3 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 841ab14..7ab8e2c 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -69,10 +69,10 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); - if (s3c2412_i2s.iis_cclk == NULL) { + if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); iounmap(s3c2412_i2s.regs); - return -ENODEV; + return PTR_ERR(s3c2412_i2s.iis_cclk); } /* Set MPLL as the source for IIS CLK */ diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 63d8849..21c92e2 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -383,10 +383,10 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) return -ENXIO; s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); - if (s3c24xx_i2s.iis_clk == NULL) { + if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); - return -ENODEV; + return PTR_ERR(s3c24xx_i2s.iis_clk); } clk_enable(s3c24xx_i2s.iis_clk); diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index dc9d551..65c1cfd 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -66,17 +66,17 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) pr_debug("%s %d\n", __func__, clk_users); if (clk_users == 0) { xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); - if (!xtal) { + if (IS_ERR(xtal)) { printk(KERN_ERR "%s cannot get xtal\n", __func__); - ret = -EBUSY; + ret = PTR_ERR(xtal); } else { pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, "pclk"); - if (!pclk) { + if (IS_ERR(pclk)) { printk(KERN_ERR "%s cannot get pclk\n", __func__); clk_put(xtal); - ret = -EBUSY; + ret = PTR_ERR(pclk); } } if (!ret) { -- 1.7.4.1

2 1

Re: [alsa-devel] [linux-next] ASoC: sound/soc/samsung/neo1973_wm8753.c build failure
by Mark Brown 16 Sep '11

16 Sep '11

On Thu, Sep 15, 2011 at 08:16:30PM +0100, Russell King wrote: > It sounds like you're in the perfect place to send them a warning > that something needs to happen on this, one way or the other. Lars-Peter is the main guy here and he already replied on this thread.

1 0

[alsa-devel] [PATCH] ASoC: Fix WM8996 DC servo operation without IRQ
by Mark Brown 15 Sep '11

15 Sep '11

We need to count the timneout down. Reported-by: Axel Lin <axel.lin(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 9 +++------ 1 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 0fed436..5ca9d76 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -719,7 +719,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) { struct i2c_client *i2c = to_i2c_client(codec->dev); struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); - int i, ret; + int ret; unsigned long timeout = 200; snd_soc_write(codec, WM8996_DC_SERVO_2, mask); @@ -734,15 +734,12 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) } else { msleep(1); - if (--i) { - timeout = 0; - break; - } + timeout--; } ret = snd_soc_read(codec, WM8996_DC_SERVO_2); dev_dbg(codec->dev, "DC servo state: %x\n", ret); - } while (ret & mask); + } while (timeout && ret & mask); if (timeout == 0) dev_err(codec->dev, "DC servo timed out for %x\n", mask); -- 1.7.6.3

2 1

[alsa-devel] snd_array_free - snd: bad kfree when HDA reconfig
by Raymond Yau 15 Sep '11

15 Sep '11

These errors "snd: bad kfree" occur when reconfig HDA from model "6stack-dig-fp" to "auto" and vice versa Is "HD-Audio reconfiguration" still experimental ? BTW, it seem that HDA Digital PCBeep assigned to different input after each reconfig input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input7 input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input8 input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input9 aplay -l card 1: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Intel [HDA Intel], device 2: AD198x Headphone [AD198x Headphone] Subdevices: 1/1 Subdevice #0: subdevice #0 echo "auto" > /sys/class/sound/hwC1D0/modelname echo 1 > /sys/class/sound/hwC1D0/reconfig aplay -l card 1: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 echo "6stack-dig-fp" > /sys/class/sound/hwC1D0/modelname echo 1 > /sys/class/sound/hwC1D0/reconfig aplay -l card 1: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Intel [HDA Intel], device 2: AD198x Headphone [AD198x Headphone] Subdevices: 1/1 Subdevice #0: subdevice #0 ALSA hda_hwdep.c:220: hda-codec: reconfiguring snd: bad kfree (called from f8be4636) Pid: 8079, comm: bash Not tainted 2.6.27.41-170.2.117.fc10.i686 #1 [<c06aa30b>] ? printk+0xf/0x14 [<f895523e>] snd_hidden_kfree+0x2a/0x70 [snd] [<f8be4636>] ? snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be4636>] snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be807e>] snd_hda_ctls_clear+0x35/0x45 [snd_hda_codec] [<f8be8180>] snd_hda_codec_reset+0xf2/0x250 [snd_hda_codec] [<f8959ab8>] ? __snd_printk+0x33/0x3c [snd] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<f8bec6bf>] reconfig_store+0x3e/0xab [snd_hda_codec] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<c0594ee1>] dev_attr_store+0x1b/0x23 [<c04ce68a>] sysfs_write_file+0xb4/0xdf [<c04ce5d6>] ? sysfs_write_file+0x0/0xdf [<c04932ee>] vfs_write+0x84/0xdf [<c04933e2>] sys_write+0x3b/0x60 [<c0404c8a>] syscall_call+0x7/0xb ======================= snd: bad kfree (called from f8be4636) Pid: 8079, comm: bash Not tainted 2.6.27.41-170.2.117.fc10.i686 #1 [<c06aa30b>] ? printk+0xf/0x14 [<f895523e>] snd_hidden_kfree+0x2a/0x70 [snd] [<f8be4636>] ? snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be4636>] snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be8089>] snd_hda_ctls_clear+0x40/0x45 [snd_hda_codec] [<f8be8180>] snd_hda_codec_reset+0xf2/0x250 [snd_hda_codec] [<f8959ab8>] ? __snd_printk+0x33/0x3c [snd] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<f8bec6bf>] reconfig_store+0x3e/0xab [snd_hda_codec] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<c0594ee1>] dev_attr_store+0x1b/0x23 [<c04ce68a>] sysfs_write_file+0xb4/0xdf [<c04ce5d6>] ? sysfs_write_file+0x0/0xdf [<c04932ee>] vfs_write+0x84/0xdf [<c04933e2>] sys_write+0x3b/0x60 [<c0404c8a>] syscall_call+0x7/0xb ======================= snd: bad kfree (called from f8be4636) Pid: 8079, comm: bash Not tainted 2.6.27.41-170.2.117.fc10.i686 #1 [<c06aa30b>] ? printk+0xf/0x14 [<f895523e>] snd_hidden_kfree+0x2a/0x70 [snd] [<f8be4636>] ? snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be4636>] snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be81de>] snd_hda_codec_reset+0x150/0x250 [snd_hda_codec] [<f8959ab8>] ? __snd_printk+0x33/0x3c [snd] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<f8bec6bf>] reconfig_store+0x3e/0xab [snd_hda_codec] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<c0594ee1>] dev_attr_store+0x1b/0x23 [<c04ce68a>] sysfs_write_file+0xb4/0xdf [<c04ce5d6>] ? sysfs_write_file+0x0/0xdf [<c04932ee>] vfs_write+0x84/0xdf [<c04933e2>] sys_write+0x3b/0x60 [<c0404c8a>] syscall_call+0x7/0xb ======================= snd: bad kfree (called from f8be4636) Pid: 8079, comm: bash Not tainted 2.6.27.41-170.2.117.fc10.i686 #1 [<c06aa30b>] ? printk+0xf/0x14 [<f895523e>] snd_hidden_kfree+0x2a/0x70 [snd] [<f8be4636>] ? snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be4636>] snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be81e9>] snd_hda_codec_reset+0x15b/0x250 [snd_hda_codec] [<f8959ab8>] ? __snd_printk+0x33/0x3c [snd] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<f8bec6bf>] reconfig_store+0x3e/0xab [snd_hda_codec] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<c0594ee1>] dev_attr_store+0x1b/0x23 [<c04ce68a>] sysfs_write_file+0xb4/0xdf [<c04ce5d6>] ? sysfs_write_file+0x0/0xdf [<c04932ee>] vfs_write+0x84/0xdf [<c04933e2>] sys_write+0x3b/0x60 [<c0404c8a>] syscall_call+0x7/0xb ======================= ALSA hda_codec.c:3961: hda_codec: model 'auto' is selected ALSA hda_codec.c:4900: autoconfig: line_outs=4 (0x12/0x16/0x24/0x25/0x0) type:line ALSA hda_codec.c:4904: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA hda_codec.c:4908: hp_outs=1 (0x11/0x0/0x0/0x0/0x0) ALSA hda_codec.c:4909: mono: mono_out=0x0 ALSA hda_codec.c:4912: dig-out=0x1b/0x0 ALSA hda_codec.c:4913: inputs:<7>ALSA hda_codec.c:4917: Front Mic=0x14<7>ALSA hda_codec.c:4917: Rear Mic=0x17<7>ALSA hda_codec.c:4917: Line=0x15<7>ALSA hda_codec.c:4917: CD=0x18<7>ALSA hda_codec.c:4919: input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input8 ALSA hda_codec.c:2370: Cannot find slave Mono Playback Volume, skipped ALSA hda_codec.c:2370: Cannot find slave Speaker Playback Volume, skipped ALSA hda_codec.c:2370: Cannot find slave Mono Playback Switch, skipped ALSA hda_codec.c:2370: Cannot find slave Speaker Playback Switch, skipped snd: bad kfree (called from f8be4636) Pid: 8079, comm: bash Not tainted 2.6.27.41-170.2.117.fc10.i686 #1 [<c06aa30b>] ? printk+0xf/0x14 [<f895523e>] snd_hidden_kfree+0x2a/0x70 [snd] [<f8be4636>] ? snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be4636>] snd_array_free+0xe/0x25 [snd_hda_codec] [<f8b79d29>] ad198x_free_kctls+0x37/0x3c [snd_hda_codec_analog] [<f8b7ad5a>] ad198x_build_controls+0x190/0x217 [snd_hda_codec_analog] [<f8be7481>] snd_hda_codec_build_controls+0x31/0x3d [snd_hda_codec] [<f8bec704>] reconfig_store+0x83/0xab [snd_hda_codec] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<c0594ee1>] dev_attr_store+0x1b/0x23 [<c04ce68a>] sysfs_write_file+0xb4/0xdf [<c04ce5d6>] ? sysfs_write_file+0x0/0xdf [<c04932ee>] vfs_write+0x84/0xdf [<c04933e2>] sys_write+0x3b/0x60 [<c0404c8a>] syscall_call+0x7/0xb ======================= ALSA hda_hwdep.c:220: hda-codec: reconfiguring snd: bad kfree (called from f8be4636) Pid: 8201, comm: bash Not tainted 2.6.27.41-170.2.117.fc10.i686 #1 [<c06aa30b>] ? printk+0xf/0x14 [<f895523e>] snd_hidden_kfree+0x2a/0x70 [snd] [<f8be4636>] ? snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be4636>] snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be807e>] snd_hda_ctls_clear+0x35/0x45 [snd_hda_codec] [<f8be8180>] snd_hda_codec_reset+0xf2/0x250 [snd_hda_codec] [<f8959ab8>] ? __snd_printk+0x33/0x3c [snd] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<f8bec6bf>] reconfig_store+0x3e/0xab [snd_hda_codec] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<c0594ee1>] dev_attr_store+0x1b/0x23 [<c04ce68a>] sysfs_write_file+0xb4/0xdf [<c04ce5d6>] ? sysfs_write_file+0x0/0xdf [<c04932ee>] vfs_write+0x84/0xdf [<c04933e2>] sys_write+0x3b/0x60 [<c0404c8a>] syscall_call+0x7/0xb ======================= snd: bad kfree (called from f8be4636) Pid: 8201, comm: bash Not tainted 2.6.27.41-170.2.117.fc10.i686 #1 [<c06aa30b>] ? printk+0xf/0x14 [<f895523e>] snd_hidden_kfree+0x2a/0x70 [snd] [<f8be4636>] ? snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be4636>] snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be8089>] snd_hda_ctls_clear+0x40/0x45 [snd_hda_codec] [<f8be8180>] snd_hda_codec_reset+0xf2/0x250 [snd_hda_codec] [<f8959ab8>] ? __snd_printk+0x33/0x3c [snd] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<f8bec6bf>] reconfig_store+0x3e/0xab [snd_hda_codec] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<c0594ee1>] dev_attr_store+0x1b/0x23 [<c04ce68a>] sysfs_write_file+0xb4/0xdf [<c04ce5d6>] ? sysfs_write_file+0x0/0xdf [<c04932ee>] vfs_write+0x84/0xdf [<c04933e2>] sys_write+0x3b/0x60 [<c0404c8a>] syscall_call+0x7/0xb ======================= snd: bad kfree (called from f8be4636) Pid: 8201, comm: bash Not tainted 2.6.27.41-170.2.117.fc10.i686 #1 [<c06aa30b>] ? printk+0xf/0x14 [<f895523e>] snd_hidden_kfree+0x2a/0x70 [snd] [<f8be4636>] ? snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be4636>] snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be81de>] snd_hda_codec_reset+0x150/0x250 [snd_hda_codec] [<f8959ab8>] ? __snd_printk+0x33/0x3c [snd] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<f8bec6bf>] reconfig_store+0x3e/0xab [snd_hda_codec] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<c0594ee1>] dev_attr_store+0x1b/0x23 [<c04ce68a>] sysfs_write_file+0xb4/0xdf [<c04ce5d6>] ? sysfs_write_file+0x0/0xdf [<c04932ee>] vfs_write+0x84/0xdf [<c04933e2>] sys_write+0x3b/0x60 [<c0404c8a>] syscall_call+0x7/0xb ======================= snd: bad kfree (called from f8be4636) Pid: 8201, comm: bash Not tainted 2.6.27.41-170.2.117.fc10.i686 #1 [<c06aa30b>] ? printk+0xf/0x14 [<f895523e>] snd_hidden_kfree+0x2a/0x70 [snd] [<f8be4636>] ? snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be4636>] snd_array_free+0xe/0x25 [snd_hda_codec] [<f8be81e9>] snd_hda_codec_reset+0x15b/0x250 [snd_hda_codec] [<f8959ab8>] ? __snd_printk+0x33/0x3c [snd] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<f8bec6bf>] reconfig_store+0x3e/0xab [snd_hda_codec] [<f8bec681>] ? reconfig_store+0x0/0xab [snd_hda_codec] [<c0594ee1>] dev_attr_store+0x1b/0x23 [<c04ce68a>] sysfs_write_file+0xb4/0xdf [<c04ce5d6>] ? sysfs_write_file+0x0/0xdf [<c04932ee>] vfs_write+0x84/0xdf [<c04933e2>] sys_write+0x3b/0x60 [<c0404c8a>] syscall_call+0x7/0xb ======================= ALSA hda_codec.c:3961: hda_codec: model '6stack-dig-fp' is selected input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input9

2 5

[alsa-devel] QasMixer 0.14.0 released
by Sebastian H. 15 Sep '11

15 Sep '11

QasMixer version 0.14.0 is now available. QasMixer is an ALSA mixer with a customized Qt GUI. Changes: * Cairo painting replaced with native Qt painting * Reduced flicker on a mixer change Homepage with more information http://xwmw.org/qasmixer Project page http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] ASoC: uninitialized variable i used in wait_for_dc_servo
by Axel Lin 15 Sep '11

15 Sep '11

hi Mark, I guess we need to initialize i to timeout value. comments? diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e5e4607..5aee9cb 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -713,6 +713,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) snd_soc_write(codec, WM8996_DC_SERVO_2, mask); /* Use the interrupt if possible */ + i = timeout; do { if (i2c->irq) { timeout = wait_for_completion_timeout(&wm8996->dcs_done, @@ -722,7 +723,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, u16 mask) } else { msleep(1); - if (--i) { + if (--i == 0) { timeout = 0; break; }

2 2

[alsa-devel] [PATCH] sound: Fix race condition in the pcm_lib "wait for space loop
by Arjan van de Ven 15 Sep '11

15 Sep '11

From 2e37f0a4b2289962e1a45d8e02f8a7f7adad619f Mon Sep 17 00:00:00 2001 From: Arjan van de Ven <arjan(a)linux.intel.com> Date: Mon, 5 Sep 2011 09:40:18 -0700 Subject: [PATCH] sound: Fix race condition in the pcm_lib "wait for space" loop The wait_for_avail() function in pcm_lib.c has a race in it (observed in practice by an Intel validation group). The function is supposed to return once space in the buffer has become available, or if some timeout happens. The entity that creates space (irq handler of sound driver and some such) will do a wake up on a waitqueue that this function registers for. However there are two races in the existing code 1) If space became available between the caller noticing there was no space and this function actually sleeping, the wakeup is missed and the timeout condition will happen instead 2) If a wakeup happened but not sufficient space became available, the code will loop again and wait for more space. However, if the second wake comes in prior to hitting the schedule_timeout_interruptible(), it will be missed, and potentially you'll wait out until the timeout happens. The fix consists of using more careful setting of the current state (so that if a wakeup happens in the main loop window, the schedule_timeout() falls through) and by checking for available space prior to going into the schedule_timeout() loop, but after being on the waitqueue and having the state set to interruptible. Signed-off-by: Arjan van de Ven <arjan(a)linux.intel.com> CC: Jaroslav Kysela <perex(a)perex.cz> CC: Takashi Iwai <tiwai(a)suse.de> CC: alsa-devel(a)alsa-project.org CC: linux-kernel(a)vger.kernel.org --- sound/core/pcm_lib.c | 29 ++++++++++++++++++++++++++--- 1 files changed, 26 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 86d0caf..8848080 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1761,6 +1761,10 @@ static int wait_for_avail(struct snd_pcm_substream *substream, snd_pcm_uframes_t avail = 0; long wait_time, tout; + init_waitqueue_entry(&wait, current); + add_wait_queue(&runtime->tsleep, &wait); + set_current_state(TASK_INTERRUPTIBLE); + if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { @@ -1771,16 +1775,34 @@ static int wait_for_avail(struct snd_pcm_substream *substream, } wait_time = msecs_to_jiffies(wait_time * 1000); } - init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->tsleep, &wait); + + /* + * We need to check if space became available already (and thus the + * wakeup happened already) prior to going into the sleep loop to + * close the race of space already having become available. + * This check must happen after been added to the waitqueue and + * having current state be INTERRUPTIBLE. + */ + + if (is_playback) + avail = snd_pcm_playback_avail(runtime); + else + avail = snd_pcm_capture_avail(runtime); + if (avail >= runtime->twake) + goto _endloop; + + for (;;) { if (signal_pending(current)) { err = -ERESTARTSYS; break; } snd_pcm_stream_unlock_irq(substream); - tout = schedule_timeout_interruptible(wait_time); + + tout = schedule_timeout(wait_time); + snd_pcm_stream_lock_irq(substream); + set_current_state(TASK_INTERRUPTIBLE); switch (runtime->status->state) { case SNDRV_PCM_STATE_SUSPENDED: err = -ESTRPIPE; @@ -1814,6 +1836,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream, break; } _endloop: + set_current_state(TASK_RUNNING); remove_wait_queue(&runtime->tsleep, &wait); *availp = avail; return err; -- 1.7.6 -- Arjan van de Ven Intel Open Source Technology Centre For development, discussion and tips for power savings, visit http://www.lesswatts.org

3 4

[alsa-devel] [PATCH] ASoC: samsung: Fix checking return value of clk_get
by Axel Lin 15 Sep '11

15 Sep '11

clk_get() returns a pointer to the struct clk or an ERR_PTR(). Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Cc: Christian Pellegrin <chripell(a)evolware.org> Cc: Ben Dooks <ben(a)simtec.co.uk> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Jassi Brar <jassisinghbrar(a)gmail.com> --- sound/soc/samsung/s3c2412-i2s.c | 2 +- sound/soc/samsung/s3c24xx-i2s.c | 2 +- sound/soc/samsung/s3c24xx_uda134x.c | 4 ++-- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 841ab14..cecdbad 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -69,7 +69,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); - if (s3c2412_i2s.iis_cclk == NULL) { + if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); iounmap(s3c2412_i2s.regs); return -ENODEV; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index 63d8849..9f888bc 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -383,7 +383,7 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) return -ENXIO; s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); - if (s3c24xx_i2s.iis_clk == NULL) { + if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); iounmap(s3c24xx_i2s.regs); return -ENODEV; diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index dc9d551..4da57cd 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -66,13 +66,13 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) pr_debug("%s %d\n", __func__, clk_users); if (clk_users == 0) { xtal = clk_get(&s3c24xx_uda134x_snd_device->dev, "xtal"); - if (!xtal) { + if (IS_ERR(xtal)) { printk(KERN_ERR "%s cannot get xtal\n", __func__); ret = -EBUSY; } else { pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, "pclk"); - if (!pclk) { + if (IS_ERR(pclk)) { printk(KERN_ERR "%s cannot get pclk\n", __func__); clk_put(xtal); -- 1.7.4.1

4 4

[alsa-devel] OSS emulation and hardware configuration
by Mike Crowe 15 Sep '11

15 Sep '11

Hi folks, Apologies if this is obvious, but how does the OSS emulation layer set hardware parameters at the codec level? We are working in an embedded environment, and using a custom codec driver. When we access it via the ALSA API, it works fine. We have a bf5xx_wm8990_hw_params() function with configures the critical hardware parameters of the codec, and I have diagnostics code in the driver which does kernel prints when the driver changes key codec parameters (such as bit rate, timing, etc). When my OSS application opens it's device and changes the audio rate, I don't see that function getting called. In core/oss/pcm_oss.c, I see the snd_pcm_oss_set_rate() function getting called, but I don't understand how it changes parameters in the codec. My bf5xx_wm8990_hw_params() never gets called. Can somebody please point me in the right direction? I have two applications who need to access the audio driver, and at different bit rates. I have to makes sure they are getting set correctly. Mike

5 9

[alsa-devel] pthread or snd_async_add_pcm_handler
by Maarten de Boer 14 Sep '11

14 Sep '11

Hello, I am doing MMAP-based full duplex audio processing. I looked at the direct_loop code in the alsa-lib/test/pcm.c example. Now I want to run do the audio processing asynchronously. The async_direct_loop in the example does this, but I was wondering: is there any advantage to using snd_async_add_pcm_handler over using a simple pthread? Thanks, Maarten

2 1

[alsa-devel] [PATCH 0/4] Cleanups for snd-usb-audio
by Daniel Mack 14 Sep '11

14 Sep '11

Hi, I prepared a subset of my work on the generic USB audio driver that doesn't actually change any logic but just moves around major chunks of code. Can we get this merged as preparation for the more intrusive parts that follow? Thanks, Daniel Daniel Mack (4): USB: Add endpoint usage definitions to ch9.h ALSA: snd-usb: re-order the Makefile ALSA: snd-usb: re-order code ALSA: snd-usb: move code from urb.c to endpoint.c include/linux/usb/ch9.h | 17 +- sound/usb/Makefile | 12 +- sound/usb/card.c | 4 +- sound/usb/endpoint.c | 1185 +++++++++++++++++++++++++++++++++-------------- sound/usb/endpoint.h | 20 +- sound/usb/pcm.c | 2 +- sound/usb/quirks.c | 7 +- sound/usb/stream.c | 452 ++++++++++++++++++ sound/usb/stream.h | 12 + sound/usb/urb.c | 941 ------------------------------------- sound/usb/urb.h | 21 - 11 files changed, 1341 insertions(+), 1332 deletions(-) create mode 100644 sound/usb/stream.c create mode 100644 sound/usb/stream.h delete mode 100644 sound/usb/urb.c delete mode 100644 sound/usb/urb.h -- 1.7.5.4

4 13

[alsa-devel] QasMixer 0.13.0 released
by Sebastian H. 14 Sep '11

14 Sep '11

QasMixer version 0.13.0 is now available. QasMixer is an ALSA mixer with a customized Qt GUI. Changes: * Minimize to tray removed - fixes disappearing issue in Gnome * Balloon style tweaks * Balloon closes on click * Balloon icon recognizes wheel events (changes the volume) * Mouse over tray icon raises the balloon * View type toolbar changed to tab selection * Playback/capture toolbar buttons moved to the bottom switches area * Improved widget focus restoring after a channel join/split * Device selection reflects the actual ALSA tree * Simplified settings dialog * Mixer window restores its former size on startup * Overhauled widget painting now uses Cairo (http://cairographics.org) * Multithreaded widget painting (if cairo version supports it) * More graphics reuse for a smaller memory footprint Homepage with more information http://xwmw.org/qasmixer Project page http://sourceforge.net/projects/qasmixer/ Happy volume changing! -- Sebastian Holtermann

1 0

[alsa-devel] [PATCH] ALSA: HDA: Cirrus - fix "Surround Speaker" volume control, name
by David Henningsson 14 Sep '11

14 Sep '11

Commit b4dabfc452a10 fixed only the "Switch" name, not the "Volume" name. -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 6

[PATCH] ALSA: HDA: Cirrus - fix "Surround Speaker" volume control name
by David Henningsson 14 Sep '11

14 Sep '11

Commit b4dabfc452a10 fixed only the "Switch" name, not the "Volume" name. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_cirrus.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d6c93d9..c45f3e6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -535,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name, int index, unsigned int pval, int dir, struct snd_kcontrol **kctlp) { - char tmp[32]; + char tmp[44]; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT); knew.private_value = pval; -- 1.7.5.4 --------------060907070702020008020606--

1 0

[PATCH] ALSA: HDA: Cirrus - fix "Surround Speaker" volume control name
by David Henningsson 14 Sep '11

14 Sep '11

This patch fixes "Surround Speaker Playback Volume" being cut off. (Commit b4dabfc452a10 was probably meant to fix this, but it fixed only the "Switch" name, not the "Volume" name.) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_cirrus.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d6c93d9..c45f3e6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -535,7 +535,7 @@ static int add_volume(struct hda_codec *codec, const char *name, int index, unsigned int pval, int dir, struct snd_kcontrol **kctlp) { - char tmp[32]; + char tmp[44]; struct snd_kcontrol_new knew = HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT); knew.private_value = pval; -- 1.7.5.4 --------------030204040609000407000502--

1 0

[alsa-devel] Player Thread is not woken after period elapsed
by Nallasellan, Singaravelan 14 Sep '11

14 Sep '11

Hi, When I tried to root cause a glitch in the audio playback, I found a weird behavior. User thread which invokes the writei function which in turn invokes a kernel function which waits for the free buffer to write the audio data. This kernel function adds this thread to a wake(sleep) queue and calls a schedule_timeout (msecs_to_jiffies(10000)). When the audio playback of a period is completed, the hardware generates an interrupt. The handler in the path wake_up the thread added to the sleep queue. Most of the time, the playback thread is woken up. During this error case, the thread is not getting woken up. However the sleep queue is not empty. This continues until the playback enter into underrun due to all the periods in the buffer are played back. Audio playback continues after recovery from xrun. Will you provide some hint on how to go about identifying the root cause? Thanks -Sing

3 7

Re: [alsa-devel] [RFC] [PATCH 02/62] mpu401:snd_mpu401_uart_new(): split semantic of irq_flags
by Clemens Ladisch 14 Sep '11

14 Sep '11

Yong Zhang wrote: > On Thu, Sep 08, 2011 at 10:27:43PM +0200, Clemens Ladisch wrote: > > I'll write a patch that changes this function so that irq<0 means > > "already reserved", and that "no interrupt" gets another bit in the > > info_flags parameter. > > So my patch will be based on yours; Here you are; any IRQF changes should now be trivial. Regards, Clemens --8<---------------------------------------------------------------->8-- [PATCH] ALSA: mpu401: clean up interrupt specification The semantics of snd_mpu401_uart_new()'s interrupt parameters are somewhat counterintuitive: To prevent the function from allocating its own interrupt, either the irq number must be invalid, or the irq_flags parameter must be zero. At the same time, the irq parameter being invalid specifies that the mpu401 code has to work without an interrupt allocated by the caller. This implies that, if there is an interrupt and it is allocated by the caller, the irq parameter must be set to a valid-looking number which then isn't actually used. With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value, which forces us to handle the parameters differently. This patch introduces a new flag MPU401_INFO_RX_POLL for interrupt-less operation, and makes the decision to allocate the interrupt to depend only on the irq parameter. Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> --- Documentation/DocBook/writing-an-alsa-driver.tmpl | 20 ++++++++++++-------- include/sound/mpu401.h | 4 ++-- sound/drivers/mpu401/mpu401.c | 6 +++--- sound/drivers/mpu401/mpu401_uart.c | 13 ++++++------- sound/isa/ad1816a/ad1816a.c | 5 +++-- sound/isa/als100.c | 10 +++++++--- sound/isa/azt2320.c | 3 ++- sound/isa/cmi8330.c | 3 ++- sound/isa/cs423x/cs4231.c | 9 ++++++--- sound/isa/cs423x/cs4236.c | 9 ++++++--- sound/isa/es18xx.c | 2 +- sound/isa/galaxy/galaxy.c | 3 ++- sound/isa/gus/gusextreme.c | 3 ++- sound/isa/opl3sa2.c | 2 +- sound/isa/sb/jazz16.c | 10 +++++++--- sound/isa/sb/sb16.c | 2 +- sound/isa/sc6000.c | 7 +++++-- sound/isa/wavefront/wavefront.c | 5 ++++- sound/pci/als4000.c | 2 +- sound/pci/au88x0/au88x0_mpu401.c | 4 ++-- sound/pci/azt3328.c | 2 +- sound/pci/cmipci.c | 2 +- sound/pci/es1938.c | 2 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 2 +- sound/pci/ice1712/ice1712.c | 4 ++-- sound/pci/maestro3.c | 2 +- sound/pci/oxygen/oxygen_lib.c | 2 +- sound/pci/riptide/riptide.c | 2 +- sound/pci/sonicvibes.c | 2 +- sound/pci/trident/trident.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/ymfpci/ymfpci.c | 2 +- 33 files changed, 89 insertions(+), 61 deletions(-) diff --git a/Documentation/DocBook/writing-an-alsa-driver.tmpl b/Documentation/DocBook/writing-an-alsa-driver.tmpl index 598c22f..24e21d0 100644 --- a/Documentation/DocBook/writing-an-alsa-driver.tmpl +++ b/Documentation/DocBook/writing-an-alsa-driver.tmpl @@ -4343,6 +4343,12 @@ struct _snd_pcm_runtime { by itself to start processing the output stream in the irq handler. </para> + <para> + If the card doesn't generate UART interrupts, add the + <constant>MPU401_INFO_RX_POLL</constant> flag. Then a timer + interrupt will be invoked for polling. + </para> + <para> Usually, the port address corresponds to the command port and port + 1 corresponds to the data port. If not, you may change @@ -4375,14 +4381,12 @@ struct _snd_pcm_runtime { </para> <para> - The 6th argument specifies the irq number for UART. If the irq - is already allocated, pass 0 to the 7th argument - (<parameter>irq_flags</parameter>). Otherwise, pass the flags - for irq allocation - (<constant>SA_XXX</constant> bits) to it, and the irq will be - reserved by the mpu401-uart layer. If the card doesn't generate - UART interrupts, pass -1 as the irq number. Then a timer - interrupt will be invoked for polling. + The 6th argument specifies the irq number that will be allocated. + Pass the irq flags for calling <function>request_irq()</function> + (<constant>IRQF_XXX</constant> bits) in the 7th argument. + If no interrupt is to be allocated (because your code is already + allocating it, or because the device does not use interrupts), + pass -1 and 0 instead. </para> </section> diff --git a/include/sound/mpu401.h b/include/sound/mpu401.h index 1f1d53f..43e24b2 100644 --- a/include/sound/mpu401.h +++ b/include/sound/mpu401.h @@ -50,6 +50,7 @@ #define MPU401_INFO_INTEGRATED (1 << 2) /* integrated h/w port */ #define MPU401_INFO_MMIO (1 << 3) /* MMIO access */ #define MPU401_INFO_TX_IRQ (1 << 4) /* independent TX irq */ +#define MPU401_INFO_RX_POLL (1 << 5) /* no RX interrupt; use timer */ #define MPU401_INFO_NO_ACK (1 << 6) /* No ACK cmd needed */ #define MPU401_MODE_BIT_INPUT 0 @@ -73,8 +74,7 @@ struct snd_mpu401 { unsigned long port; /* base port of MPU-401 chip */ unsigned long cport; /* port + 1 (usually) */ struct resource *res; /* port resource */ - int irq; /* IRQ number of MPU-401 chip (-1 = poll) */ + int irq; /* IRQ number of MPU-401 chip */ - int irq_flags; unsigned long mode; /* MPU401_MODE_XXXX */ int timer_invoked; diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c index 149d05a..4917ea1 100644 --- a/sound/drivers/mpu401/mpu401.c +++ b/sound/drivers/mpu401/mpu401.c @@ -85,9 +85,9 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard) strcat(card->longname, "polled"); } - err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], 0, - irq[dev], irq[dev] >= 0 ? IRQF_DISABLED : 0, - NULL); + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port[dev], + irq[dev] >= 0 ? 0 : MPU401_INFO_RX_POLL, + irq[dev], IRQF_DISABLED, NULL); if (err < 0) { printk(KERN_ERR "MPU401 not detected at 0x%lx\n", port[dev]); goto _err; diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 2af0999..5642f8f 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -3,7 +3,7 @@ * Routines for control of MPU-401 in UART mode * * MPU-401 supports UART mode which is not capable generate transmit - * interrupts thus output is done via polling. Also, if irq < 0, then + * interrupts thus output is done via polling. Without interrupt, * input is done also via polling. Do not expect good performance. * * @@ -374,7 +374,7 @@ snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up) /* first time - flush FIFO */ while (max-- > 0) mpu->read(mpu, MPU401D(mpu)); - if (mpu->irq < 0) + if (mpu->info_flags & MPU401_INFO_RX_POLL) snd_mpu401_uart_add_timer(mpu, 1); } @@ -383,7 +383,7 @@ snd_mpu401_uart_input_trigger(struct snd_rawmidi_substream *substream, int up) snd_mpu401_uart_input_read(mpu); spin_unlock_irqrestore(&mpu->input_lock, flags); } else { - if (mpu->irq < 0) + if (mpu->info_flags & MPU401_INFO_RX_POLL) snd_mpu401_uart_remove_timer(mpu, 1); clear_bit(MPU401_MODE_BIT_INPUT_TRIGGER, &mpu->mode); } @@ -496,7 +496,7 @@ static struct snd_rawmidi_ops snd_mpu401_uart_input = static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi) { struct snd_mpu401 *mpu = rmidi->private_data; - if (mpu->irq_flags && mpu->irq >= 0) + if (mpu->irq >= 0) free_irq(mpu->irq, (void *) mpu); release_and_free_resource(mpu->res); kfree(mpu); @@ -510,7 +510,7 @@ static void snd_mpu401_uart_free(struct snd_rawmidi *rmidi) * @port: the base address of MPU401 port * @info_flags: bitflags MPU401_INFO_XXX * @irq: the irq number, -1 if no interrupt for mpu - * @irq_flags: the irq request flags (SA_XXX), 0 if irq was already reserved. + * @irq_flags: the irq request flags (IRQF_XXX) * @rrawmidi: the pointer to store the new rawmidi instance * * Creates a new MPU-401 instance. @@ -577,7 +577,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, mpu->cport = port + 2; else mpu->cport = port + 1; - if (irq >= 0 && irq_flags) { + if (irq >= 0) { if (request_irq(irq, snd_mpu401_uart_interrupt, irq_flags, "MPU401 UART", (void *) mpu)) { snd_printk(KERN_ERR "mpu401_uart: " @@ -588,7 +588,6 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, } mpu->info_flags = info_flags; mpu->irq = irq; - mpu->irq_flags = irq_flags; if (card->shortname[0]) snprintf(rmidi->name, sizeof(rmidi->name), "%s MIDI", card->shortname); diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 3cb75bc..490d51f 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -203,9 +203,10 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard } if (mpu_port[dev] > 0) { + int mpu_flags = mpu_irq[dev] >= 0 ? 0 : MPU401_INFO_RX_POLL; if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED, - NULL) < 0) + mpu_port[dev], mpu_flags, + mpu_irq[dev], IRQF_DISABLED, NULL) < 0) printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpu_port[dev]); } diff --git a/sound/isa/als100.c b/sound/isa/als100.c index 20becc8..3355609 100644 --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -245,18 +245,22 @@ static int __devinit snd_card_als100_probe(int dev, if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { int mpu_type = MPU401_HW_ALS100; + int mpu_flags = 0; - if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) { mpu_irq[dev] = -1; + mpu_flags = MPU401_INFO_RX_POLL; + } if (pid->driver_data == SB_HW_DT019X) mpu_type = MPU401_HW_MPU401; if (snd_mpu401_uart_new(card, 0, mpu_type, - mpu_port[dev], 0, + mpu_port[dev], + mpu_flags, mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + IRQF_DISABLED, NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); } diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c index aac8dc1..4dc2dc9 100644 --- a/sound/isa/azt2320.c +++ b/sound/isa/azt2320.c @@ -232,8 +232,9 @@ static int __devinit snd_card_azt2320_probe(int dev, } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { + int mpu_flags = mpu_irq[dev] >= 0 ? 0 : MPU401_INFO_RX_POLL; if (snd_mpu401_uart_new(card, 0, MPU401_HW_AZT2320, - mpu_port[dev], 0, + mpu_port[dev], mpu_flags, mpu_irq[dev], IRQF_DISABLED, NULL) < 0) snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]); diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index fe79a16..7829aaa 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -595,8 +595,9 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) } if (mpuport[dev] != SNDRV_AUTO_PORT) { + int mpuflags = mpuirq[dev] >= 0 ? 0 : MPU401_INFO_RX_POLL; if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpuport[dev], 0, mpuirq[dev], + mpuport[dev], mpuflags, mpuirq[dev], IRQF_DISABLED, NULL) < 0) printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", mpuport[dev]); diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c index cb9153e..6ae72ea 100644 --- a/sound/isa/cs423x/cs4231.c +++ b/sound/isa/cs423x/cs4231.c @@ -127,11 +127,14 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n) goto out; if (mpu_port[n] > 0 && mpu_port[n] != SNDRV_AUTO_PORT) { + int mpu_flags = 0; - if (mpu_irq[n] == SNDRV_AUTO_IRQ) + if (mpu_irq[n] == SNDRV_AUTO_IRQ) { mpu_irq[n] = -1; + mpu_flags = MPU401_INFO_RX_POLL; + } if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232, - mpu_port[n], 0, mpu_irq[n], - mpu_irq[n] >= 0 ? IRQF_DISABLED : 0, + mpu_port[n], mpu_flags, mpu_irq[n], + IRQF_DISABLED, NULL) < 0) dev_warn(dev, "MPU401 not detected\n"); } diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index 999dc1e..71994df 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -445,12 +445,15 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { + int mpu_flags = 0; - if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) { mpu_irq[dev] = -1; + mpu_flags = MPU401_INFO_RX_POLL; + } if (snd_mpu401_uart_new(card, 0, MPU401_HW_CS4232, - mpu_port[dev], 0, + mpu_port[dev], mpu_flags, mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, NULL) < 0) + IRQF_DISABLED, NULL) < 0) printk(KERN_WARNING IDENT ": MPU401 not detected\n"); } diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index fb4d6b3..106760f 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2161,7 +2161,7 @@ static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev) if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { err = snd_mpu401_uart_new(card, 0, MPU401_HW_ES18XX, mpu_port[dev], 0, - irq[dev], 0, &chip->rmidi); + -1, 0, &chip->rmidi); if (err < 0) return err; } diff --git a/sound/isa/galaxy/galaxy.c b/sound/isa/galaxy/galaxy.c index ee54df0..645deff 100644 --- a/sound/isa/galaxy/galaxy.c +++ b/sound/isa/galaxy/galaxy.c @@ -584,8 +584,9 @@ static int __devinit snd_galaxy_probe(struct device *dev, unsigned int n) goto error; if (mpu_port[n] >= 0) { + int mpu_flags = mpu_irq[n] >= 0 ? 0 : MPU401_INFO_RX_POLL; err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port[n], 0, mpu_irq[n], + mpu_port[n], mpu_flags, mpu_irq[n], IRQF_DISABLED, NULL); if (err < 0) goto error; diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c index 008e8e5..872b659 100644 --- a/sound/isa/gus/gusextreme.c +++ b/sound/isa/gus/gusextreme.c @@ -316,8 +316,9 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n) } if (es1688->mpu_port >= 0x300) { + int mpu_flags = mpu_irq[n] >= 0 ? 0 : MPU401_INFO_RX_POLL; error = snd_mpu401_uart_new(card, 0, MPU401_HW_ES1688, - es1688->mpu_port, 0, + es1688->mpu_port, mpu_flags, mpu_irq[n], IRQF_DISABLED, NULL); if (error < 0) goto out; diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 9b915e2..38dc53b 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -708,7 +708,7 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev) if (midi_port[dev] >= 0x300 && midi_port[dev] < 0x340) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_OPL3SA2, midi_port[dev], 0, - xirq, 0, &chip->rmidi)) < 0) + -1, 0, &chip->rmidi)) < 0) return err; } sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d", diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 8ccbcdd..49ee7db 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -315,14 +315,18 @@ static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev) goto err_free; } if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) { + int mpu_flags = 0; + - if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) { mpu_irq[dev] = -1; + mpu_flags = MPU401_INFO_RX_POLL; + } if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port[dev], 0, + mpu_port[dev], mpu_flags, mpu_irq[dev], - mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + IRQF_DISABLED, NULL) < 0) snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n", mpu_port[dev]); diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 4d1c5a3..8d751ba 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -395,7 +395,7 @@ static int __devinit snd_sb16_probe(struct snd_card *card, int dev) if (chip->mpu_port > 0 && chip->mpu_port != SNDRV_AUTO_PORT) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SB, chip->mpu_port, 0, - xirq, 0, &chip->rmidi)) < 0) + -1, 0, &chip->rmidi)) < 0) return err; chip->rmidi_callback = snd_mpu401_uart_interrupt; } diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 9a8bbf6..6f778cf 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -653,11 +653,14 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) } if (mpu_port[dev] != SNDRV_AUTO_PORT) { + int mpu_flags = 0; - if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) { mpu_irq[dev] = -1; + mpu_flags = MPU401_INFO_RX_POLL; + } if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, - mpu_port[dev], 0, + mpu_port[dev], mpu_flags, mpu_irq[dev], IRQF_DISABLED, NULL) < 0) snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n", diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 711670e..dc2548f 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -447,8 +447,11 @@ snd_wavefront_probe (struct snd_card *card, int dev) /* -------- CS4232 MPU-401 interface -------- */ if (cs4232_mpu_port[dev] > 0 && cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) { + int mpu_flags = 0; + if (cs4232_mpu_irq[dev] < 0) + mpu_flags = MPU401_INFO_RX_POLL; err = snd_mpu401_uart_new(card, midi_dev, MPU401_HW_CS4232, - cs4232_mpu_port[dev], 0, + cs4232_mpu_port[dev], mpu_flags, cs4232_mpu_irq[dev], IRQF_DISABLED, NULL); if (err < 0) { diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index a9c1af3..8f65858 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -932,7 +932,7 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, if ((err = snd_mpu401_uart_new( card, 0, MPU401_HW_ALS4000, iobase + ALS4K_IOB_30_MIDI_DATA, MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi)) < 0) { + -1, 0, &chip->rmidi)) < 0) { printk(KERN_ERR "als4000: no MPU-401 device at 0x%lx?\n", iobase + ALS4K_IOB_30_MIDI_DATA); goto out_err; diff --git a/sound/pci/au88x0/au88x0_mpu401.c b/sound/pci/au88x0/au88x0_mpu401.c index 0dc8d25..344d329 100644 --- a/sound/pci/au88x0/au88x0_mpu401.c +++ b/sound/pci/au88x0/au88x0_mpu401.c @@ -84,7 +84,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) #ifdef VORTEX_MPU401_LEGACY if ((temp = snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_MPU401, 0x330, - 0, 0, 0, &rmidi)) != 0) { + 0, -1, 0, &rmidi)) != 0) { hwwrite(vortex->mmio, VORTEX_CTRL, (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN); @@ -95,7 +95,7 @@ static int __devinit snd_vortex_midi(vortex_t * vortex) if ((temp = snd_mpu401_uart_new(vortex->card, 0, MPU401_HW_AUREAL, port, MPU401_INFO_INTEGRATED | MPU401_INFO_MMIO, - 0, 0, &rmidi)) != 0) { + -1, 0, &rmidi)) != 0) { hwwrite(vortex->mmio, VORTEX_CTRL, (hwread(vortex->mmio, VORTEX_CTRL) & ~CTRL_MIDI_PORT) & ~CTRL_MIDI_EN); diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 579fc0d..9c5461a 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -2653,7 +2653,7 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) err = snd_mpu401_uart_new( card, 0, MPU401_HW_AZT2320, chip->mpu_io, MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rmidi + -1, 0, &chip->rmidi ); if (err < 0) { snd_printk(KERN_ERR "azf3328: no MPU-401 device at 0x%lx?\n", diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 9cf99fb..5e89278 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -3229,7 +3229,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc iomidi, (integrated_midi ? MPU401_INFO_INTEGRATED : 0), - cm->irq, 0, &cm->rmidi)) < 0) { + -1, 0, &cm->rmidi)) < 0) { printk(KERN_ERR "cmipci: no UART401 device at 0x%lx\n", iomidi); } } diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 26a5a2f..09d3af6 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1855,7 +1855,7 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci, } if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, chip->mpu_port, MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi) < 0) { + -1, 0, &chip->rmidi) < 0) { printk(KERN_ERR "es1938: unable to initialize MPU-401\n"); } else { // this line is vital for MIDI interrupt handling on ess-solo1 diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index 99ea932..16d897b 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2844,7 +2844,7 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci, if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, chip->io_port + ESM_MPU401_PORT, MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi)) < 0) { + -1, 0, &chip->rmidi)) < 0) { printk(KERN_WARNING "es1968: skipping MPU-401 MIDI support..\n"); } } diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index f9123f0..746568e 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1307,7 +1307,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_FM801, FM801_REG(chip, MPU401_DATA), MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi)) < 0) { + -1, 0, &chip->rmidi)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 0ccc0eb..051fc08 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2749,7 +2749,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, ICEREG(ice, MPU1_CTRL), (c->mpu401_1_info_flags | MPU401_INFO_INTEGRATED), - ice->irq, 0, &ice->rmidi[0]); + -1, 0, &ice->rmidi[0]); if (err < 0) { snd_card_free(card); return err; @@ -2765,7 +2765,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, err = snd_mpu401_uart_new(card, 1, MPU401_HW_ICE1712, ICEREG(ice, MPU2_CTRL), (c->mpu401_2_info_flags | MPU401_INFO_INTEGRATED), - ice->irq, 0, &ice->rmidi[1]); + -1, 0, &ice->rmidi[1]); if (err < 0) { snd_card_free(card); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 0378126..9c8dd1d 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2821,7 +2821,7 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) err = snd_mpu401_uart_new(chip->card, 0, MPU401_HW_MPU401, chip->iobase + MPU401_DATA_PORT, MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi); + -1, 0, &chip->rmidi); if (err < 0) printk(KERN_WARNING "maestro3: no MIDI support.\n"); #endif diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 82311fc..4882c62 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -685,7 +685,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, info_flags |= MPU401_INFO_INPUT; err = snd_mpu401_uart_new(card, 0, MPU401_HW_CMIPCI, chip->addr + OXYGEN_MPU401, - info_flags, 0, 0, + info_flags, -1, 0, &chip->midi); if (err < 0) goto err_card; diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e34ae14..1af64a1 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -2109,7 +2109,7 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) val = mpu_port[dev]; pci_write_config_word(chip->pci, PCI_EXT_MPU_Base, val); err = snd_mpu401_uart_new(card, 0, MPU401_HW_RIPTIDE, - val, 0, chip->irq, 0, + val, 0, -1, 0, &chip->rmidi); if (err < 0) snd_printk(KERN_WARNING diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 2571a67..3da90cd 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -1494,7 +1494,7 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci, } if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_SONICVIBES, sonic->midi_port, MPU401_INFO_INTEGRATED, - sonic->irq, 0, + -1, 0, &midi_uart)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index d8a128f..8ea70b9 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -149,7 +149,7 @@ static int __devinit snd_trident_probe(struct pci_dev *pci, (err = snd_mpu401_uart_new(card, 0, MPU401_HW_TRID4DWAVE, trident->midi_port, MPU401_INFO_INTEGRATED, - trident->irq, 0, &trident->rmidi)) < 0) { + -1, 0, &trident->rmidi)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index f03fd62..7575ff8 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2069,7 +2069,7 @@ static int __devinit snd_via686_init_misc(struct via82xx *chip) if (chip->mpu_res) { if (snd_mpu401_uart_new(chip->card, 0, MPU401_HW_VIA686A, mpu_port, MPU401_INFO_INTEGRATED, - chip->irq, 0, &chip->rmidi) < 0) { + -1, 0, &chip->rmidi) < 0) { printk(KERN_WARNING "unable to initialize MPU-401" " at 0x%lx, skipping\n", mpu_port); legacy &= ~VIA_FUNC_ENABLE_MIDI; diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index 511d576..ad974fd 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -306,7 +306,7 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci, if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_YMFPCI, mpu_port[dev], MPU401_INFO_INTEGRATED, - pci->irq, 0, &chip->rawmidi)) < 0) { + -1, 0, &chip->rawmidi)) < 0) { printk(KERN_WARNING "ymfpci: cannot initialize MPU401 at 0x%lx, skipping...\n", mpu_port[dev]); legacy_ctrl &= ~YMFPCI_LEGACY_MIEN; /* disable MPU401 irq */ pci_write_config_word(pci, PCIR_DSXG_LEGACY, legacy_ctrl);

3 9

Re: [alsa-devel] Player Thread is not woken after period elapsed
by Nallasellan, Singaravelan 14 Sep '11

14 Sep '11

> > > > > Further investigation of this issue shows that the thread is in > > > > > running state > > Hmm, how do you detect it? > > > > and it is not scheduled. I am not sure how I will go about debugging this > issue. > > > Any pointer would help. > > Could 'cat /proc/sched_debug' give some useful info? Tried. It did not help. > > And BTW, have you tried CONFIG_DETECT_SOFTLOCKUP = y? > It is already set.

2 1

[alsa-devel] Dell XPS 13 (Codec: IDT 92HD73C1X5) and 3 stereo jack output
by Ruben Van Boxem 14 Sep '11

14 Sep '11

Hi, I have been trying to get my Dell XPS 13 to output 5.1 sound through the stereo jacks (because the NVIDIA HDMI only does 2.1). Attached is alsa-info output. I can't seem to be able to set the Line/Mic In to output. I have tried several things, going from fiddling with Kmix to fiddling with the modprobe "model=" parameter, and even looking through the hda_analyzer.py output. This last one lifted a tiny corner of the solution, but I don't think a vanilla Alsa install will get me through this. In Windows, I can use the three jacks to output 2 channels each, so I am sure the hardware is capable of it. In HDA analyzer I can see that there are several "Line Out" entries, but since they all have the node cap "Unsolicited Capabilities", I'm pretty sure the problem is alsa not knowing about the, well, capabilities. I searched the web and documentation for anything sane to try, but nothing worked. The default modprobe "model=" is the best (I can get 4.0 output from the jacks, and everything works, except the third jack). If you need more input or information, please ask, I'm very willing to get this solved! Thanks, Ruben

2 3

Re: [alsa-devel] [PATCH] ASoC: playpaq_wm8510: Return proper error if clk_get fails
by Girdwood, Liam 13 Sep '11

13 Sep '11

On 9 September 2011 03:15, Axel Lin <axel.lin(a)gmail.com> wrote: > Return proper error instead of 0 if clk_get fails. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/atmel/playpaq_wm8510.c | 5 ++++- > 1 files changed, 4 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/atmel/playpaq_wm8510.c > b/sound/soc/atmel/playpaq_wm8510.c > index 1aac2f4..2909bfa 100644 > --- a/sound/soc/atmel/playpaq_wm8510.c > +++ b/sound/soc/atmel/playpaq_wm8510.c > @@ -383,14 +383,17 @@ static int __init playpaq_asoc_init(void) > _gclk0 = clk_get(NULL, "gclk0"); > if (IS_ERR(_gclk0)) { > _gclk0 = NULL; > + ret = PTR_ERR(_gclk0); > goto err_gclk0; > } > _pll0 = clk_get(NULL, "pll0"); > if (IS_ERR(_pll0)) { > _pll0 = NULL; > + ret = PTR_ERR(_pll0); > goto err_pll0; > } > - if (clk_set_parent(_gclk0, _pll0)) { > + ret = clk_set_parent(_gclk0, _pll0); > + if (ret) { > pr_warning("snd-soc-playpaq: " > "Failed to set PLL0 as parent for DAC clock\n"); > goto err_set_clk; > -- > 1.7.4.1 > > > > Acked-by: Liam Girdwood <lrg(a)ti.com>

2 1

[alsa-devel] [PATCH 1/2] regulator: Implement deferred disable support
by Mark Brown 13 Sep '11

13 Sep '11

It is a reasonably common pattern for hardware to require some delay after being quiesced before the disable has finalised, especially in mixed signal devices. For example, an active discharge may be required to ensure that the circuit starts up again in a known state. Avoid having to implement such delays in the regulator API by providing regulator_deferred_disable() which will do a regulator_disable() a specified number of milliseconds after it is called. Due to the reference counting done on regulators a deferred disable can be cancelled by doing another regulator_enable(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/regulator/core.c | 59 ++++++++++++++++++++++++++++++++++++ include/linux/regulator/consumer.h | 7 ++++ include/linux/regulator/driver.h | 3 ++ 3 files changed, 69 insertions(+), 0 deletions(-) diff --git a/drivers/regulator/core.c b/drivers/regulator/core.c index 9a33fe2..026f3d4 100644 --- a/drivers/regulator/core.c +++ b/drivers/regulator/core.c @@ -1552,6 +1552,63 @@ int regulator_force_disable(struct regulator *regulator) } EXPORT_SYMBOL_GPL(regulator_force_disable); +static void regulator_disable_work(struct work_struct *work) +{ + struct regulator_dev *rdev = container_of(work, struct regulator_dev, + disable_work.work); + int count, i, ret; + + mutex_lock(&rdev->mutex); + + BUG_ON(!rdev->deferred_disables); + + count = rdev->deferred_disables; + rdev->deferred_disables = 0; + + for (i = 0; i < count; i++) { + ret = _regulator_disable(rdev); + if (ret != 0) + rdev_err(rdev, "Deferred disable failed: %d\n", ret); + } + + mutex_unlock(&rdev->mutex); + + if (rdev->supply) { + for (i = 0; i < count; i++) { + ret = regulator_disable(rdev->supply); + if (ret != 0) { + rdev_err(rdev, + "Supply disable failed: %d\n", ret); + } + } + } +} + +/** + * regulator_disable_deferred - disable regulator output with delay + * @regulator: regulator source + * @ms: miliseconds until the regulator is disabled + * + * Execute regulator_disable() on the regulator after a delay. This + * is intended for use with devices that require some time to quiesce. + * + * NOTE: this will only disable the regulator output if no other consumer + * devices have it enabled, the regulator device supports disabling and + * machine constraints permit this operation. + */ +int regulator_disable_deferred(struct regulator *regulator, int ms) +{ + struct regulator_dev *rdev = regulator->rdev; + + mutex_lock(&rdev->mutex); + rdev->deferred_disables++; + mutex_unlock(&rdev->mutex); + + return schedule_delayed_work(&rdev->disable_work, + msecs_to_jiffies(ms)); +} +EXPORT_SYMBOL_GPL(regulator_disable_deferred); + static int _regulator_is_enabled(struct regulator_dev *rdev) { /* If we don't know then assume that the regulator is always on */ @@ -2622,6 +2679,7 @@ struct regulator_dev *regulator_register(struct regulator_desc *regulator_desc, INIT_LIST_HEAD(&rdev->consumer_list); INIT_LIST_HEAD(&rdev->list); BLOCKING_INIT_NOTIFIER_HEAD(&rdev->notifier); + INIT_DELAYED_WORK(&rdev->disable_work, regulator_disable_work); /* preform any regulator specific init */ if (init_data->regulator_init) { @@ -2729,6 +2787,7 @@ void regulator_unregister(struct regulator_dev *rdev) #ifdef CONFIG_DEBUG_FS debugfs_remove_recursive(rdev->debugfs); #endif + flush_work_sync(&rdev->disable_work.work); WARN_ON(rdev->open_count); unset_regulator_supplies(rdev); list_del(&rdev->list); diff --git a/include/linux/regulator/consumer.h b/include/linux/regulator/consumer.h index b47771a..f7756d1 100644 --- a/include/linux/regulator/consumer.h +++ b/include/linux/regulator/consumer.h @@ -141,6 +141,7 @@ int regulator_enable(struct regulator *regulator); int regulator_disable(struct regulator *regulator); int regulator_force_disable(struct regulator *regulator); int regulator_is_enabled(struct regulator *regulator); +int regulator_disable_deferred(struct regulator *regulator, int ms); int regulator_bulk_get(struct device *dev, int num_consumers, struct regulator_bulk_data *consumers); @@ -211,6 +212,12 @@ static inline int regulator_disable(struct regulator *regulator) return 0; } +static inline int regulator_disable_deferred(struct regulator *regulator, + int ms) +{ + return 0; +} + static inline int regulator_is_enabled(struct regulator *regulator) { return 1; diff --git a/include/linux/regulator/driver.h b/include/linux/regulator/driver.h index 03a1798..52c89ae 100644 --- a/include/linux/regulator/driver.h +++ b/include/linux/regulator/driver.h @@ -200,6 +200,9 @@ struct regulator_dev { struct regulation_constraints *constraints; struct regulator *supply; /* for tree */ + struct delayed_work disable_work; + int deferred_disables; + void *reg_data; /* regulator_dev data */ #ifdef CONFIG_DEBUG_FS -- 1.7.5.4

2 4

[alsa-devel] ASoC updates in 3.1
by Mark Brown 13 Sep '11

13 Sep '11

The following changes since commit 18036b5866b5e407a28f444a80de186a5d7df767: ASoC: Correct element count for WM8996 sidetone HPF (2011-08-24 17:36:12 +0100) Added Arnd's fix. are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc.git for-3.1 Arnd Bergmann (1): ASoC: Fix trivial build regression in Kirkwood I2S Fabio Estevam (1): ASoC: imx: Fix build warning of unused 'card' variable Lars-Peter Clausen (3): ASoC: snd_soc_codec_{readable,writable}_register change default to true ASoC: Fix register cache sync register_writable WARN_ONs ASoC: Blackfin: bf5xx-ad193x: Fix codec device name Mark Brown (2): MAINTAINERS: Add some missed Wolfson files ASoC: Fix reporting of partial jack updates Peter Ujfalusi (1): ASoC: soc-dapm: Fix parameter comment for snd_soc_dapm_free Timur Tabi (1): ASoC: MPC5200: replace of_device with platform_device MAINTAINERS | 4 ++++ sound/soc/blackfin/bf5xx-ad193x.c | 4 ++-- sound/soc/fsl/mpc5200_dma.c | 6 +++--- sound/soc/imx/imx-pcm-fiq.c | 1 - sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/soc-cache.c | 12 ++++++------ sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-jack.c | 2 +- 9 files changed, 20 insertions(+), 17 deletions(-)

2 1

[alsa-devel] Determining the plugins being used?
by Raymond Toy 13 Sep '11

13 Sep '11

Is there a way to determine the plugins being used? I notice that snd_pcm_dump produces information about the various plugins that are being used. In one of test case, aplay -v shows that it's using a rate conversion plugin, a slave for Direct Stream Mixing PCM, and finally, the actual hardware PCM card. For my application, it would be nice to know if we're using the pulse audio plugin so that I can setup our code appropriately. (The period interrupts are not what we were expecting, so we need to use a different code path.) If we're using the dmix plugin, the period interrupts can match our expectations, so we want to use a low-latency path. What is the appropriate way to determine the plugins used? Ray

2 3

Re: [alsa-devel] ASoC MPC5xxx PSC AC97 audio driver
by jonsmirl＠gmail.com 13 Sep '11

13 Sep '11

On Thu, Sep 8, 2011 at 6:45 AM, David Jander <david.jander(a)protonic.nl> wrote: > > Dear Jon, > > Thanks for replying so quickly. > > On Thu, 8 Sep 2011 06:28:02 -0400 > "jonsmirl(a)gmail.com" <jonsmirl(a)gmail.com> wrote: >> On Thu, Sep 8, 2011 at 6:16 AM, David Jander <david.jander(a)protonic.nl> >> wrote: >> > >> > Dear Jon, >> >> Here's the device tree... >> http://git.digispeaker.com/?p=digispeaker-kernel.git;a=blob;f=arch/powerpc/… > > Thanks... but it uses I2S, not AC97 :-( > It doesn't have a "fsl,mpc5200-pcm" node either, btw. > In the case of I2C, this makes some sense, since the Codec is usually > connected to two different interfaces at the same time (I2S and I2C for > control), so you have 2 device nodes. AC97 OTOH has just one interface. It is > connected to a AC97 bus (a PSC in this case). AFAICS there are no OF bindings > yet for an AC97 bus, so the actual codec doesn't figure and thus still needs > this ugly fabric driver thing. Try the pcm030 device tree... http://git.digispeaker.com/?p=digispeaker-kernel.git;a=blob;f=arch/powerpc/… I think it is checked into the kernel. > >> Grant can comment on the current bindings. > > Ok. > > Best regards, > > -- > David Jander > Protonic Holland. > -- Jon Smirl jonsmirl(a)gmail.com

5 15

[alsa-devel] Getting the actual HW sample rate
by Raymond Toy 12 Sep '11

12 Sep '11

I know that http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg24794.htmlgi… code to get the available sample rates (among other things). However, my quest is to get the actual HW sample rate. This code is quite helpful. I know that if a sound is already playing, I can look at /proc/asound/card0/pcm0p/sub0/hw_params to see what the HW sample rate is. But is there a way to get this using ALSA? In my application, we want to know the actual HW rate so that we can do the sample rate conversion ourselves upfront and therefore we want to avoid yet another sample rate conversion when playing out the audio. (The application can synthesize audio and we can select the sample rate for the synthesis.) Thanks, Ray

2 5

[alsa-devel] [PATCH] ALSA ymfpci - Add "Playback" to FM Legacy Volume control
by Raymond Yau 12 Sep '11

12 Sep '11

Add "Playback" to "FM Legacy Volume" control since - YDSXGR_LEGACYOUTVOL is a Playback Volume control for OPL3 FM Synth

2 3

Re: [alsa-devel] [PATCH v2] ALSA: aoa: Remove obsolete cleanup for clientdata
by Takashi Iwai 12 Sep '11

12 Sep '11

At Fri, 09 Sep 2011 19:04:45 +0800, Axel Lin wrote: > > The i2c core will clear the clientdata pointer automatically. > We don't have to set the `data' field to NULL in remove() or > if probe() failed anymore. > > Also remove a unneeded NULL checking for kfree. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > Reviewed-by: Wolfram Sang <w.sang(a)pengutronix.de> Applied now. Thanks. Takashi > --- > sound/aoa/codecs/onyx.c | 4 +--- > 1 files changed, 1 insertions(+), 3 deletions(-) > > diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c > index 3687a6c..762af68 100644 > --- a/sound/aoa/codecs/onyx.c > +++ b/sound/aoa/codecs/onyx.c > @@ -1067,7 +1067,6 @@ static int onyx_i2c_probe(struct i2c_client *client, > printk(KERN_DEBUG PFX "created and attached onyx instance\n"); > return 0; > fail: > - i2c_set_clientdata(client, NULL); > kfree(onyx); > return -ENODEV; > } > @@ -1112,8 +1111,7 @@ static int onyx_i2c_remove(struct i2c_client *client) > > aoa_codec_unregister(&onyx->codec); > of_node_put(onyx->codec.node); > - if (onyx->codec_info) > - kfree(onyx->codec_info); > + kfree(onyx->codec_info); > kfree(onyx); > return 0; > } > -- > 1.7.4.1 > > >

1 0

Re: [alsa-devel] [PATCH 2/2] ALSA: keywest: Remove obsolete cleanup for clientdata
by Takashi Iwai 12 Sep '11

12 Sep '11

At Fri, 09 Sep 2011 17:50:52 +0800, Axel Lin wrote: > > The i2c core will clear the clientdata pointer automatically. > We don't have to set the `data' field to NULL in remove() or > if probe() failed anymore. > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied now. Thanks. Takashi > --- > sound/ppc/keywest.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c > index 8f064c7..4080bec 100644 > --- a/sound/ppc/keywest.c > +++ b/sound/ppc/keywest.c > @@ -82,7 +82,6 @@ static int keywest_attach_adapter(struct i2c_adapter *adapter) > > static int keywest_remove(struct i2c_client *client) > { > - i2c_set_clientdata(client, NULL); > if (! keywest_ctx) > return 0; > if (client == keywest_ctx->client) > -- > 1.7.4.1 > > >

1 0

[alsa-devel] [PATCH 1/1] HDA - Add Headphone Playback Volume control for ad1988/ad1989
by Raymond Yau 12 Sep '11

12 Sep '11

HDA - Add Headphone Playback Volume control for ad1988/ad1989 - use DAC0 instead of DAC1 for Port-A Headphone - assign 0x03 to spec->multiout.hp_nid except model="6stack-dig-fp"

2 1

[alsa-devel] [PATCH v4] ASoC: omap: convert per-board modules to platform drivers
by Mans Rullgard 12 Sep '11

12 Sep '11

This converts the per-board modules to platform drivers for a device created by in main platform setup. These drivers call snd_soc_register_card() directly instead of going via a "soc-audio" device and the corresponding driver in soc-core. Signed-off-by: Mans Rullgard <mans.rullgard(a)linaro.org> --- Removed the unnecessary machine_is_foo() checks from the board drivers. --- arch/arm/mach-omap2/devices.c | 34 ++++++++++++++++++++ sound/soc/omap/am3517evm.c | 52 ++++++++++++++++++------------ sound/soc/omap/igep0020.c | 49 ++++++++++++++++++---------- sound/soc/omap/n810.c | 70 +++++++++++++++++++++++++---------------- sound/soc/omap/omap3beagle.c | 52 ++++++++++++++++++------------ sound/soc/omap/omap3evm.c | 53 +++++++++++++++++++------------ sound/soc/omap/omap3pandora.c | 67 +++++++++++++++++++++++---------------- sound/soc/omap/overo.c | 53 ++++++++++++++++++------------ sound/soc/omap/rx51.c | 52 ++++++++++++++++++++---------- sound/soc/omap/sdp3430.c | 62 +++++++++++++++++++++-------------- sound/soc/omap/sdp4430.c | 57 ++++++++++++++++++++------------- sound/soc/omap/zoom2.c | 65 ++++++++++++++++++++++++------------- 12 files changed, 425 insertions(+), 241 deletions(-) diff --git a/arch/arm/mach-omap2/devices.c b/arch/arm/mach-omap2/devices.c index 5b8ca68..cfec01b 100644 --- a/arch/arm/mach-omap2/devices.c +++ b/arch/arm/mach-omap2/devices.c @@ -299,6 +299,10 @@ static struct platform_device omap_pcm = { .id = -1, }; +static struct platform_device omap_soc_audio = { + .id = -1, +}; + /* * OMAP2420 has 2 McBSP ports * OMAP2430 has 5 McBSP ports @@ -311,8 +315,30 @@ OMAP_MCBSP_PLATFORM_DEVICE(3); OMAP_MCBSP_PLATFORM_DEVICE(4); OMAP_MCBSP_PLATFORM_DEVICE(5); +static struct { + int machine; + const char *name; +} soc_device_names[] = { + { MACH_TYPE_OMAP3517EVM, "am3517evm-soc-audio" }, + { MACH_TYPE_IGEP0020, "igep2-soc-audio" }, + { MACH_TYPE_NOKIA_N810, "n8x1-soc-audio" }, + { MACH_TYPE_NOKIA_N810_WIMAX, "n8x1-soc-audio" }, + { MACH_TYPE_OMAP3_BEAGLE, "omap3beagle-soc-audio" }, + { MACH_TYPE_DEVKIT8000, "omap3beagle-soc-audio" }, + { MACH_TYPE_OMAP3EVM, "omap3evm-soc-audio" }, + { MACH_TYPE_OMAP3_PANDORA, "pandora-soc-audio" }, + { MACH_TYPE_OVERO, "overo-soc-audio", }, + { MACH_TYPE_CM_T35, "overo-soc-audio", }, + { MACH_TYPE_NOKIA_RX51, "rx51-soc-audio", }, + { MACH_TYPE_OMAP_3430SDP, "sdp3430-soc-audio", }, + { MACH_TYPE_OMAP_4430SDP, "sdp4430-soc-audio", }, + { MACH_TYPE_OMAP_ZOOM2, "zoom2-soc-audio", }, +}; + static void omap_init_audio(void) { + int i; + platform_device_register(&omap_mcbsp1); platform_device_register(&omap_mcbsp2); if (cpu_is_omap243x() || cpu_is_omap34xx() || cpu_is_omap44xx()) { @@ -323,6 +349,14 @@ static void omap_init_audio(void) platform_device_register(&omap_mcbsp5); platform_device_register(&omap_pcm); + + for (i = 0; i < ARRAY_SIZE(soc_device_names); i++) { + if (machine_arch_type == soc_device_names[i].machine) { + omap_soc_audio.name = soc_device_names[i].name; + platform_device_register(&omap_soc_audio); + break; + } + } } #else diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c index 73dde4a..1624885 100644 --- a/sound/soc/omap/am3517evm.c +++ b/sound/soc/omap/am3517evm.c @@ -151,45 +151,57 @@ static struct snd_soc_card snd_soc_am3517evm = { .num_links = 1, }; -static struct platform_device *am3517evm_snd_device; - -static int __init am3517evm_soc_init(void) +static int __devinit am3517evm_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_am3517evm; int ret; - if (!machine_is_omap3517evm()) - return -ENODEV; pr_info("OMAP3517 / AM3517 EVM SoC init\n"); - am3517evm_snd_device = platform_device_alloc("soc-audio", -1); - if (!am3517evm_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; } - platform_set_drvdata(am3517evm_snd_device, &snd_soc_am3517evm); + return 0; +} + +static int __devexit am3517evm_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); - ret = platform_device_add(am3517evm_snd_device); - if (ret) - goto err1; + snd_soc_unregister_card(card); return 0; +} + +static struct platform_driver am3517evm_driver = { + .driver = { + .name = "am3517evm-soc-audio", + .owner = THIS_MODULE, + }, -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(am3517evm_snd_device); + .probe = am3517evm_soc_probe, + .remove = __devexit_p(am3517evm_soc_remove), +}; - return ret; +static int __init am3517evm_soc_init(void) +{ + return platform_driver_register(&am3517evm_driver); } +module_init(am3517evm_soc_init); static void __exit am3517evm_soc_exit(void) { - platform_device_unregister(am3517evm_snd_device); + platform_driver_unregister(&am3517evm_driver); } - -module_init(am3517evm_soc_init); module_exit(am3517evm_soc_exit); MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:am3517evm-soc-audio"); diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c index 0ae3470..6056558 100644 --- a/sound/soc/omap/igep0020.c +++ b/sound/soc/omap/igep0020.c @@ -94,44 +94,57 @@ static struct snd_soc_card snd_soc_card_igep2 = { .num_links = 1, }; -static struct platform_device *igep2_snd_device; - -static int __init igep2_soc_init(void) +static int __devinit igep2_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_card_igep2; int ret; - if (!machine_is_igep0020()) - return -ENODEV; printk(KERN_INFO "IGEP v2 SoC init\n"); - igep2_snd_device = platform_device_alloc("soc-audio", -1); - if (!igep2_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; } - platform_set_drvdata(igep2_snd_device, &snd_soc_card_igep2); + return 0; +} - ret = platform_device_add(igep2_snd_device); - if (ret) - goto err1; +static int __devexit igep2_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); return 0; +} + +static struct platform_driver igep2_driver = { + .driver = { + .name = "igep2-soc-audio", + .owner = THIS_MODULE, + }, -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(igep2_snd_device); + .probe = igep2_soc_probe, + .remove = __devexit_p(igep2_soc_remove), +}; - return ret; +static int __init igep2_soc_init(void) +{ + return platform_driver_register(&igep2_driver); } module_init(igep2_soc_init); static void __exit igep2_soc_exit(void) { - platform_device_unregister(igep2_snd_device); + platform_driver_unregister(&igep2_driver); } module_exit(igep2_soc_exit); MODULE_AUTHOR("Enric Balletbo i Serra <eballetbo(a)iseebcn.com>"); MODULE_DESCRIPTION("ALSA SoC IGEP v2"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:igep2-soc-audio"); diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 83d213b..6e8ba6a 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -323,38 +323,25 @@ static struct snd_soc_card snd_soc_n810 = { .num_links = 1, }; -static struct platform_device *n810_snd_device; - -static int __init n810_soc_init(void) +static int __devinit n810_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_n810; + struct device *dev = &pdev->dev; int err; - struct device *dev; - - if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) - return -ENODEV; - n810_snd_device = platform_device_alloc("soc-audio", -1); - if (!n810_snd_device) - return -ENOMEM; - - platform_set_drvdata(n810_snd_device, &snd_soc_n810); - err = platform_device_add(n810_snd_device); - if (err) - goto err1; - - dev = &n810_snd_device->dev; + card->dev = dev; sys_clkout2_src = clk_get(dev, "sys_clkout2_src"); if (IS_ERR(sys_clkout2_src)) { dev_err(dev, "Could not get sys_clkout2_src clock\n"); err = PTR_ERR(sys_clkout2_src); - goto err2; + return err; } sys_clkout2 = clk_get(dev, "sys_clkout2"); if (IS_ERR(sys_clkout2)) { dev_err(dev, "Could not get sys_clkout2\n"); err = PTR_ERR(sys_clkout2); - goto err3; + goto err1; } /* * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use @@ -364,7 +351,7 @@ static int __init n810_soc_init(void) if (IS_ERR(func96m_clk)) { dev_err(dev, "Could not get func 96M clock\n"); err = PTR_ERR(func96m_clk); - goto err4; + goto err2; } clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); @@ -375,33 +362,62 @@ static int __init n810_soc_init(void) gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); + err = snd_soc_register_card(card); + if (err) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + err); + goto err3; + } + return 0; -err4: - clk_put(sys_clkout2); err3: - clk_put(sys_clkout2_src); + clk_put(func96m_clk); err2: - platform_device_del(n810_snd_device); + clk_put(sys_clkout2); err1: - platform_device_put(n810_snd_device); + clk_put(sys_clkout2_src); return err; } -static void __exit n810_soc_exit(void) +static int __devexit n810_soc_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + gpio_free(N810_SPEAKER_AMP_GPIO); gpio_free(N810_HEADSET_AMP_GPIO); clk_put(sys_clkout2_src); clk_put(sys_clkout2); clk_put(func96m_clk); - platform_device_unregister(n810_snd_device); + snd_soc_unregister_card(card); + + return 0; } +static struct platform_driver n810_driver = { + .driver = { + .name = "n8x0-soc-audio", + .owner = THIS_MODULE, + }, + + .probe = n810_soc_probe, + .remove = __devexit_p(n810_soc_remove), +}; + +static int __init n810_soc_init(void) +{ + return platform_driver_register(&n810_driver); +} module_init(n810_soc_init); + +static void __exit n810_soc_exit(void) +{ + platform_driver_unregister(&n810_driver); +} module_exit(n810_soc_exit); MODULE_AUTHOR("Jarkko Nikula <jhnikula(a)gmail.com>"); MODULE_DESCRIPTION("ALSA SoC Nokia N810"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:n8x0-soc-audio"); diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index 40db813..46594a4 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -105,45 +105,57 @@ static struct snd_soc_card snd_soc_omap3beagle = { .num_links = 1, }; -static struct platform_device *omap3beagle_snd_device; - -static int __init omap3beagle_soc_init(void) +static int __devinit omap3beagle_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_omap3beagle; int ret; - if (!(machine_is_omap3_beagle() || machine_is_devkit8000())) - return -ENODEV; pr_info("OMAP3 Beagle/Devkit8000 SoC init\n"); - omap3beagle_snd_device = platform_device_alloc("soc-audio", -1); - if (!omap3beagle_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; } - platform_set_drvdata(omap3beagle_snd_device, &snd_soc_omap3beagle); + return 0; +} + +static int __devexit omap3beagle_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); - ret = platform_device_add(omap3beagle_snd_device); - if (ret) - goto err1; + snd_soc_unregister_card(card); return 0; +} + +static struct platform_driver omap3beagle_driver = { + .driver = { + .name = "omap3beagle-soc-audio", + .owner = THIS_MODULE, + }, -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(omap3beagle_snd_device); + .probe = omap3beagle_soc_probe, + .remove = __devexit_p(omap3beagle_soc_remove), +}; - return ret; +static int __init omap3beagle_soc_init(void) +{ + return platform_driver_register(&omap3beagle_driver); } +module_init(omap3beagle_soc_init); static void __exit omap3beagle_soc_exit(void) { - platform_device_unregister(omap3beagle_snd_device); + platform_driver_unregister(&omap3beagle_driver); } - -module_init(omap3beagle_soc_init); module_exit(omap3beagle_soc_exit); MODULE_AUTHOR("Steve Sakoman <steve(a)sakoman.com>"); MODULE_DESCRIPTION("ALSA SoC OMAP3 Beagle"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:omap3beagle-soc-audio"); diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c index 0daa044..9df49fc 100644 --- a/sound/soc/omap/omap3evm.c +++ b/sound/soc/omap/omap3evm.c @@ -92,44 +92,57 @@ static struct snd_soc_card snd_soc_omap3evm = { .num_links = 1, }; -static struct platform_device *omap3evm_snd_device; - -static int __init omap3evm_soc_init(void) +static int __devinit omap3evm_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_omap3evm; int ret; - if (!machine_is_omap3evm()) - return -ENODEV; pr_info("OMAP3 EVM SoC init\n"); - omap3evm_snd_device = platform_device_alloc("soc-audio", -1); - if (!omap3evm_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; } - platform_set_drvdata(omap3evm_snd_device, &snd_soc_omap3evm); - ret = platform_device_add(omap3evm_snd_device); - if (ret) - goto err1; + return 0; +} + +static int __devexit omap3evm_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); return 0; +} + +static struct platform_driver omap3evm_driver = { + .driver = { + .name = "omap3evm-soc-audio", + .owner = THIS_MODULE, + }, -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(omap3evm_snd_device); + .probe = omap3evm_soc_probe, + .remove = __devexit_p(omap3evm_soc_remove), +}; - return ret; +static int __init omap3evm_soc_init(void) +{ + return platform_driver_register(&omap3evm_driver); } +module_init(omap3evm_soc_init); static void __exit omap3evm_soc_exit(void) { - platform_device_unregister(omap3evm_snd_device); + platform_driver_unregister(&omap3evm_driver); } - -module_init(omap3evm_soc_init); module_exit(omap3evm_soc_exit); MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal(a)ti.com>"); MODULE_DESCRIPTION("ALSA SoC OMAP3 EVM"); MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:omap3evm-soc-audio"); diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 8047c52..5d369f8 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -252,17 +252,15 @@ static struct snd_soc_card snd_soc_card_omap3pandora = { .num_links = ARRAY_SIZE(omap3pandora_dai), }; -static struct platform_device *omap3pandora_snd_device; - -static int __init omap3pandora_soc_init(void) +static int __devinit omap3pandora_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_card_omap3pandora; int ret; - if (!machine_is_omap3_pandora()) - return -ENODEV; - pr_info("OMAP3 Pandora SoC init\n"); + card->dev = &pdev->dev; + ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); if (ret) { pr_err(PREFIX "Failed to get DAC power GPIO\n"); @@ -287,53 +285,68 @@ static int __init omap3pandora_soc_init(void) goto fail1; } - omap3pandora_snd_device = platform_device_alloc("soc-audio", -1); - if (omap3pandora_snd_device == NULL) { - pr_err(PREFIX "Platform device allocation failed\n"); - ret = -ENOMEM; - goto fail1; - } - - platform_set_drvdata(omap3pandora_snd_device, &snd_soc_card_omap3pandora); - - ret = platform_device_add(omap3pandora_snd_device); + ret = snd_soc_register_card(card); if (ret) { - pr_err(PREFIX "Unable to add platform device\n"); - goto fail2; + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + goto fail1; } - omap3pandora_dac_reg = regulator_get(&omap3pandora_snd_device->dev, "vcc"); + omap3pandora_dac_reg = regulator_get(card->dev, "vcc"); if (IS_ERR(omap3pandora_dac_reg)) { pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n", - dev_name(&omap3pandora_snd_device->dev), + dev_name(card->dev), PTR_ERR(omap3pandora_dac_reg)); ret = PTR_ERR(omap3pandora_dac_reg); - goto fail3; + goto fail2; } return 0; -fail3: - platform_device_del(omap3pandora_snd_device); fail2: - platform_device_put(omap3pandora_snd_device); + snd_soc_unregister_card(card); fail1: gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); fail0: gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); return ret; } -module_init(omap3pandora_soc_init); -static void __exit omap3pandora_soc_exit(void) +static int __devexit omap3pandora_soc_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + regulator_put(omap3pandora_dac_reg); - platform_device_unregister(omap3pandora_snd_device); + snd_soc_unregister_card(card); gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); + + return 0; +} + +static struct platform_driver omap3pandora_driver = { + .driver = { + .name = "pandora-soc-audio", + .owner = THIS_MODULE, + }, + + .probe = omap3pandora_soc_probe, + .remove = __devexit_p(omap3pandora_soc_remove), +}; + +static int __init omap3pandora_soc_init(void) +{ + return platform_driver_register(&omap3pandora_driver); +} +module_init(omap3pandora_soc_init); + +static void __exit omap3pandora_soc_exit(void) +{ + platform_driver_unregister(&omap3pandora_driver); } module_exit(omap3pandora_soc_exit); MODULE_AUTHOR("Grazvydas Ignotas <notasas(a)gmail.com>"); MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pandora-soc-audio"); diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index bbcf380..3a6f4ba 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -94,46 +94,57 @@ static struct snd_soc_card snd_soc_card_overo = { .num_links = 1, }; -static struct platform_device *overo_snd_device; - -static int __init overo_soc_init(void) +static int __devinit overo_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_card_overo; int ret; - if (!(machine_is_overo() || machine_is_cm_t35())) { - pr_debug("Incomatible machine!\n"); - return -ENODEV; - } - printk(KERN_INFO "overo SoC init\n"); + pr_info("overo SoC init\n"); + + card->dev = &pdev->dev; - overo_snd_device = platform_device_alloc("soc-audio", -1); - if (!overo_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; } - platform_set_drvdata(overo_snd_device, &snd_soc_card_overo); + return 0; +} + +static int __devexit overo_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); - ret = platform_device_add(overo_snd_device); - if (ret) - goto err1; + snd_soc_unregister_card(card); return 0; +} -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(overo_snd_device); +static struct platform_driver overo_driver = { + .driver = { + .name = "overo-soc-audio", + .owner = THIS_MODULE, + }, - return ret; + .probe = overo_soc_probe, + .remove = __devexit_p(overo_soc_remove), +}; + +static int __init overo_soc_init(void) +{ + return platform_driver_register(&overo_driver); } module_init(overo_soc_init); static void __exit overo_soc_exit(void) { - platform_device_unregister(overo_snd_device); + platform_driver_unregister(&overo_driver); } module_exit(overo_soc_exit); MODULE_AUTHOR("Steve Sakoman <steve(a)sakoman.com>"); MODULE_DESCRIPTION("ALSA SoC overo"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:overo-soc-audio"); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 0aae998..8cad848 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -408,14 +408,12 @@ static struct snd_soc_card rx51_sound_card = { .num_configs = ARRAY_SIZE(rx51_codec_conf), }; -static struct platform_device *rx51_snd_device; - -static int __init rx51_soc_init(void) +static int __devinit rx51_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &rx51_sound_card; int err; - if (!machine_is_nokia_rx51()) - return -ENODEV; + card->dev = &pdev->dev; err = gpio_request_one(RX51_TVOUT_SEL_GPIO, GPIOF_DIR_OUT | GPIOF_INIT_LOW, "tvout_sel"); @@ -426,21 +424,14 @@ static int __init rx51_soc_init(void) if (err) goto err_gpio_eci_sw; - rx51_snd_device = platform_device_alloc("soc-audio", -1); - if (!rx51_snd_device) { - err = -ENOMEM; + err = snd_soc_register_card(card); + if (err) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + err); goto err1; } - platform_set_drvdata(rx51_snd_device, &rx51_sound_card); - - err = platform_device_add(rx51_snd_device); - if (err) - goto err2; - return 0; -err2: - platform_device_put(rx51_snd_device); err1: gpio_free(RX51_ECI_SW_GPIO); err_gpio_eci_sw: @@ -450,19 +441,44 @@ err_gpio_tvout_sel: return err; } -static void __exit rx51_soc_exit(void) +static int __devexit rx51_soc_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios), rx51_av_jack_gpios); - platform_device_unregister(rx51_snd_device); + snd_soc_unregister_card(card); + gpio_free(RX51_ECI_SW_GPIO); gpio_free(RX51_TVOUT_SEL_GPIO); + + return 0; } +static struct platform_driver rx51_driver = { + .driver = { + .name = "rx51-soc-audio", + .owner = THIS_MODULE, + }, + + .probe = rx51_soc_probe, + .remove = __devexit_p(rx51_soc_remove), +}; + +static int __init rx51_soc_init(void) +{ + return platform_driver_register(&rx51_driver); +} module_init(rx51_soc_init); + +static void __exit rx51_soc_exit(void) +{ + platform_driver_unregister(&rx51_driver); +} module_exit(rx51_soc_exit); MODULE_AUTHOR("Nokia Corporation"); MODULE_DESCRIPTION("ALSA SoC Nokia RX-51"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:rx51-soc-audio"); diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 3f72d17..83667ce 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -289,24 +289,15 @@ static struct snd_soc_card snd_soc_sdp3430 = { .num_links = ARRAY_SIZE(sdp3430_dai), }; -static struct platform_device *sdp3430_snd_device; - -static int __init sdp3430_soc_init(void) +static int __devinit sdp3430_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_sdp3430; int ret; u8 pin_mux; - if (!machine_is_omap_3430sdp()) - return -ENODEV; - printk(KERN_INFO "SDP3430 SoC init\n"); - - sdp3430_snd_device = platform_device_alloc("soc-audio", -1); - if (!sdp3430_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } + pr_info("SDP3430 SoC init\n"); - platform_set_drvdata(sdp3430_snd_device, &snd_soc_sdp3430); + card->dev = &pdev->dev; /* Set TWL4030 GPIO6 as EXTMUTE signal */ twl_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, @@ -316,30 +307,51 @@ static int __init sdp3430_soc_init(void) twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, TWL4030_INTBR_PMBR1); - ret = platform_device_add(sdp3430_snd_device); - if (ret) - goto err1; + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } return 0; +} -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(sdp3430_snd_device); +static int __devexit sdp3430_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); - return ret; + snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), + hs_jack_gpios); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver sdp3430_driver = { + .driver = { + .name = "sdp3430-soc-audio", + .owner = THIS_MODULE, + }, + + .probe = sdp3430_soc_probe, + .remove = __devexit_p(sdp3430_soc_remove), +}; + +static int __init sdp3430_soc_init(void) +{ + return platform_driver_register(&sdp3430_driver); } module_init(sdp3430_soc_init); static void __exit sdp3430_soc_exit(void) { - snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios), - hs_jack_gpios); - - platform_device_unregister(sdp3430_snd_device); + platform_driver_unregister(&sdp3430_driver); } module_exit(sdp3430_soc_exit); MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); MODULE_DESCRIPTION("ALSA SoC SDP3430"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:sdp3430-soc-audio"); diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 189e039..ee98155 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -215,47 +215,60 @@ static struct snd_soc_card snd_soc_sdp4430 = { .num_links = 1, }; -static struct platform_device *sdp4430_snd_device; - -static int __init sdp4430_soc_init(void) +static int __devinit sdp4430_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_sdp4430; int ret; - if (!machine_is_omap_4430sdp()) - return -ENODEV; - printk(KERN_INFO "SDP4430 SoC init\n"); - - sdp4430_snd_device = platform_device_alloc("soc-audio", -1); - if (!sdp4430_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } + pr_info("SDP4430 SoC init\n"); - platform_set_drvdata(sdp4430_snd_device, &snd_soc_sdp4430); + card->dev = &pdev->dev; - ret = platform_device_add(sdp4430_snd_device); - if (ret) - goto err; + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; + } /* Codec starts in HP mode */ twl6040_power_mode = 1; return 0; +} -err: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(sdp4430_snd_device); - return ret; +static int __devexit sdp4430_soc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver sdp4430_driver = { + .driver = { + .name = "sdp4430-soc-audio", + .owner = THIS_MODULE, + }, + + .probe = sdp4430_soc_probe, + .remove = __devexit_p(sdp4430_soc_remove), +}; + +static int __init sdp4430_soc_init(void) +{ + return platform_driver_register(&sdp4430_driver); } module_init(sdp4430_soc_init); static void __exit sdp4430_soc_exit(void) { - platform_device_unregister(sdp4430_snd_device); + platform_driver_unregister(&sdp4430_driver); } module_exit(sdp4430_soc_exit); MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); MODULE_DESCRIPTION("ALSA SoC SDP4430"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:sdp4430-soc-audio"); diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c index 0170994..8406f59 100644 --- a/sound/soc/omap/zoom2.c +++ b/sound/soc/omap/zoom2.c @@ -239,26 +239,14 @@ static struct snd_soc_card snd_soc_zoom2 = { .num_links = ARRAY_SIZE(zoom2_dai), }; -static struct platform_device *zoom2_snd_device; - -static int __init zoom2_soc_init(void) +static int __devinit zoom2_soc_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &snd_soc_zoom2; int ret; - if (!machine_is_omap_zoom2()) - return -ENODEV; - printk(KERN_INFO "Zoom2 SoC init\n"); - - zoom2_snd_device = platform_device_alloc("soc-audio", -1); - if (!zoom2_snd_device) { - printk(KERN_ERR "Platform device allocation failed\n"); - return -ENOMEM; - } + pr_info("Zoom2 SoC init\n"); - platform_set_drvdata(zoom2_snd_device, &snd_soc_zoom2); - ret = platform_device_add(zoom2_snd_device); - if (ret) - goto err1; + card->dev = &pdev->dev; BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0); gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0); @@ -266,26 +254,57 @@ static int __init zoom2_soc_init(void) BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0); gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0); + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + goto err; + } + return 0; -err1: - printk(KERN_ERR "Unable to add platform device\n"); - platform_device_put(zoom2_snd_device); +err: + gpio_free(ZOOM2_HEADSET_MUX_GPIO); + gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO); return ret; } -module_init(zoom2_soc_init); -static void __exit zoom2_soc_exit(void) +static int __devexit zoom2_soc_remove(struct platform_device *pdev) { + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + gpio_free(ZOOM2_HEADSET_MUX_GPIO); gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO); - platform_device_unregister(zoom2_snd_device); + return 0; +} + +static struct platform_driver zoom2_driver = { + .driver = { + .name = "zoom2-soc-audio", + .owner = THIS_MODULE, + }, + + .probe = zoom2_soc_probe, + .remove = __devexit_p(zoom2_soc_remove), +}; + +static int __init zoom2_soc_init(void) +{ + return platform_driver_register(&zoom2_driver); +} +module_init(zoom2_soc_init); + +static void __exit zoom2_soc_exit(void) +{ + platform_driver_unregister(&zoom2_driver); } module_exit(zoom2_soc_exit); MODULE_AUTHOR("Misael Lopez Cruz <x0052729(a)ti.com>"); MODULE_DESCRIPTION("ALSA SoC Zoom2"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:zoom2-soc-audio"); -- 1.7.6.1

2 1

Re: [alsa-devel] [PATCH v2] sound: usb: mixer: increase control transfer timeout
by Jassi Brar 12 Sep '11

12 Sep '11

On Mon, Aug 29, 2011 at 6:01 PM, Felipe Balbi <balbi(a)ti.com> wrote: > 100ms might not be enough for slower devices. Increasing > timeout to 1000ms doesn't seem like overkill. > > Signed-off-by: Felipe Balbi <balbi(a)ti.com> > --- > > Changes since v1: > . Increase all 100ms timeout to 1000ms > > sound/usb/mixer.c | 4 ++-- > sound/usb/mixer_quirks.c | 10 +++++----- > 2 files changed, 7 insertions(+), 7 deletions(-) > > diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c > index cdd19d7..1c9d443 100644 > --- a/sound/usb/mixer.c > +++ b/sound/usb/mixer.c > @@ -296,7 +296,7 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v > if (snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), request, > USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, > validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), > - buf, val_len, 100) >= val_len) { > + buf, val_len, 1000) >= val_len) { > *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); > snd_usb_autosuspend(cval->mixer->chip); > return 0; > @@ -445,7 +445,7 @@ int snd_usb_mixer_set_ctl_value(struct usb_mixer_elem_info *cval, > usb_sndctrlpipe(chip->dev, 0), request, > USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_OUT, > validx, snd_usb_ctrl_intf(chip) | (cval->id << 8), > - buf, val_len, 100) >= 0) { > + buf, val_len, 1000) >= 0) { > snd_usb_autosuspend(chip); > return 0; > } > diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c > index 3d0f487..7c41399 100644 > --- a/sound/usb/mixer_quirks.c > +++ b/sound/usb/mixer_quirks.c > @@ -190,18 +190,18 @@ static int snd_audigy2nx_led_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e > err = snd_usb_ctl_msg(mixer->chip->dev, > usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, > - !value, 0, NULL, 0, 100); > + !value, 0, NULL, 0, 1000); > /* USB X-Fi S51 Pro */ > if (mixer->chip->usb_id == USB_ID(0x041e, 0x30df)) > err = snd_usb_ctl_msg(mixer->chip->dev, > usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, > - !value, 0, NULL, 0, 100); > + !value, 0, NULL, 0, 1000); > else > err = snd_usb_ctl_msg(mixer->chip->dev, > usb_sndctrlpipe(mixer->chip->dev, 0), 0x24, > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, > - value, index + 2, NULL, 0, 100); > + value, index + 2, NULL, 0, 1000); > if (err < 0) > return err; > mixer->audigy2nx_leds[index] = value; > @@ -299,7 +299,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, > usb_rcvctrlpipe(mixer->chip->dev, 0), > UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | > USB_RECIP_INTERFACE, 0, > - jacks[i].unitid << 8, buf, 3, 100); > + jacks[i].unitid << 8, buf, 3, 1000); > if (err == 3 && (buf[0] == 3 || buf[0] == 6)) > snd_iprintf(buffer, "%02x %02x\n", buf[1], buf[2]); > else > @@ -332,7 +332,7 @@ static int snd_xonar_u1_switch_put(struct snd_kcontrol *kcontrol, > err = snd_usb_ctl_msg(mixer->chip->dev, > usb_sndctrlpipe(mixer->chip->dev, 0), 0x08, > USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, > - 50, 0, &new_status, 1, 100); > + 50, 0, &new_status, 1, 1000); How about also converting to use a local define for timeout value ?

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[alsa-devel] [PATCH 1/4] sound/soc/kirkwood/kirkwood-i2s.c: add missing kfree
by Julia Lawall 12 Sep '11

12 Sep '11

From: Julia Lawall <julia(a)diku.dk> Adjust the goto to jump to the error handling code that includes kfree. A simplified version of the semantic match that finds this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ identifier x; expression E1!=0,E2,E3,E4; statement S; iterator I; @@ ( if (...) { ... when != kfree(x) when != x = E3 when != E3 = x * return ...; } ... when != x = E2 when != I(...,x,...) S if (...) { ... when != x = E4 kfree(x); ... return ...; } ) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index a33fc51..8f16cd3 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -424,7 +424,7 @@ static __devinit int kirkwood_i2s_dev_probe(struct platform_device *pdev) if (!priv->mem) { dev_err(&pdev->dev, "request_mem_region failed\n"); err = -EBUSY; - goto error; + goto error_alloc; } priv->io = ioremap(priv->mem->start, SZ_16K);

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