Changes v1.0.20 v1.0.21
From AlsaProject
Changelog between 1.0.20 and 1.0.21 releases
alsa-driver
Sound Core
- Release v1.0.2
- Add compat header for linux/regulator/consumer.
- Clean up / improve INSTALL documen
- Allow relative path to --with-moddir configure optio
- Add linux/math64.h compat heade
- Add check of linux/bug.h in configure scrip
- sound: make OSS device number claiming optional and schedule its remova
ALSA Core
- Add missing definition of KERN_DEFAULT used in misc.c for older kernel
- Add compat header for linux/regulator/consumer.
- Move the previous hack to adriver.
- Add a hack to avoid Oops related with jack laye
- Fix build of hda_intel.
- Show the stack trace at bad kfree debug message
- Add krealloc() workaround for older kernels in core/info.
- Use memdup_user() wrapper when memory-debug option is enable
- Add missing PCI_VDEVICE definition for older kernel
- Add missing const to memdup_user() wrapper in adriver.
- Add linux/math64.h compat heade
- ctxfi - Add new PCI ids to pci_ids_compat.h.i
- ALSA: Fixed a typo of printk(
- ALSA: pcm - Increase protocol versio
- ALSA: Add debug module optio
- ALSA: core - strip too long file names in snd_print*(
- ALSA: Fix SG-buffer DMA with non-coherent architecture
- ALSA: info - Use krealloc(
- ALSA: Core - clean up snd_card_set_id* calls and remove possible id collisio
- ALSA: Fix double locking of card list in snd_card_register(
- ALSA: Core - add snd_card_set_id() functio
- ALSA: clean up the logic for building sequencer module
- ALSA: PCM midlevel: improve fifo_size handlin
- ALSA: Remove deprecated include/sound/driver.
- ALSA: Remove deprecated snd_card_new(
SoC PXA2xx Core
- ASoC: Pass correct platform data from pxa2xx-ac9
- ALSA: Allow passing platform_data for pxa2xx-ac9
- ASoC: change set_tdm_slot api to allow slot_width override
- [ARM] pxa/palm: Add Palm27x aSoC driver to PalmTE
- ASoC: remove duplicated code on pxa-ssp.
- ASoC: Only disable pxa2xx-i2s clocks if we enabled the
- ASoC: pxa2xx-i2s: Fix suspend/resum
- ASoC: pxa2xx-i2s: Fix inappropriate release of i2s cloc
- ASoC: pxa2xx-i2s: Handle SACR1_DRPL and SACR1_DREC separatel
- ASoC: pxa2xx-i2s: Proper hw initializatio
- ASoC: pxa2xx-i2s: Proper initializatio
- ASoC: Enforce symmetric rates for PXA2xx I2
- ASoC: em-x270: make the driver support also eXeda and CM-X300 machine
- ASoC: IMote2 ASoC Suppor
- ASoC: change stereo/mono to 32-bit/16-bit for pxa-ss
- ASoC: simplify the SSP DMA parameters settings by run-time generatio
- ASoC: pxa-ssp.c fix clock/frame inver
Control Midlevel
- sound: snd_ctl_remove_user_ctl: prevent removal of kernel control
- sound: snd_ctl_remove_unlocked_id: simplify user control countin
- sound: snd_ctl_remove_unlocked_id: simplify error path
- sound: snd_ctl_elem_add: fix value count chec
- ALSA: Add new TLV types for dBwith min/ma
Jack Input Event Midlevel
- ALSA: use card device as parent for jack input-device
PCM Midlevel
- Refresh pcm_native.patch for drain ioctl fixe
- Regenerate pcm_native.patc
- ALSA: pcm - Fix drain behavior in non-blocking mod
- ALSA: pcm - Tell user that stream to be rewound is suspende
- sound: pcm_lib: fix unsorted list constraint handlin
- ALSA: pcm - Fix hwptr buffer-size overlap bu
- ALSA: pcm - Fix warnings in debug logging
- ALSA: pcm - Add logging of hwptr updates and interrupt update
- ALSA: pcm - Fix regressions with VMwar
- ALSA: Fix SG-buffer DMA with non-coherent architecture
- sound: fix check for return value in snd_pcm_hw_refin
- ALSA: pcm - A helper function to compose PCM stream name for debug print
- ALSA: pcm - Fix update of runtime->hw_ptr_interrup
- ALSA: Clean up 64bit division function
- ALSA: PCM midlevel: Fix hw_ptr_jiffies update commi
- ALSA: PCM midlevel: lower jiffies check margin using runtime->delay valu
- ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not change
- ALSA: PCM midlevel: introduce mask for xrun_debug() macr
- ALSA: PCM midlevel: improve fifo_size handlin
- ALSA: Enable PCM hw_ptr_jiffies check only in xrun_debug mod
- ALSA: Fix invalid jiffies check after paus
- ALSA: Add extra delay count in PC
RawMidi Midlevel
- sound: rawmidi: disable active-sensing-on-close by defaul
T5 and LifeDrive
- ASoC: Switch palm27x-asoc to jack detection ap
- [ARM] pxa/palm: Add Palm27x aSoC driver to PalmTE
/include/Makefile
- Fix mrproper make targe
/soc/Makefile
- Fix build of soc-core.c with older kernel
- ASoC: add DMA platform driver for MX1x and MX2
- ASoC: Begin to factor out register cache I/O function
- ASoC: Add TXx9 AC link controller driver (v3
- ASoC: Add driver for s6000 I2S interfac
/soc/codecs/Makefile
- ASoC: Add ak4642/ak4643 codec suppor
- ASoC: Factor out shared code from WM899
- sound: new ad1836 codec driver based on aso
- ASoC: Add WM8776 CODEC drive
- ASoC: Add WM8974 CODEC drive
- ASoC: Add support for Conexant CX20442-11 voice modem code
- ASoC: new ad1938 codec driver based on aso
- ASoC: MAX9877: add MAX9877 amp drive
- ASoC: Add WM8993 CODEC drive
- ASoC: Add WM8523 CODEC drive
- ASoC: Add WM8961 drive
- ASoC: Add dummy S/PDIF codec suppor
- ASoC: Codec for STAC9766 used on the Efik
- ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier drive
- sound: ASoC WM8940 Drive
- ASoC: Add WM8960 CODEC drive
- ASoC: Add WM8988 CODEC drive
/soc/pxa/Makefile
- ASoC: IMote2 ASoC Suppor
AC97 Codec
- ALSA: Allow passing platform_data for pxa2xx-ac9
- ALSA: Allow passing platform_data to devices attached to AC97 bu
- ALSA: Kill truncate warning by shortening Sigmatel-specific AC97 control nam
ALI5451 driver
- ALSA: ali5451: remove dead cod
- ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready(
ALSA sequencer
- ALSA: OSS sequencer should be initialized after snd_seq_system_client_ini
- sound: rawmidi: disable active-sensing-on-close by defaul
- sound: seq_midi: do not send MIDI reset when closin
- sound: seq-midi: always log message on output overru
- sound: seq_midi_event: fix decoding of (N)RPN event
- ALSA: clean up the logic for building sequencer module
ALSA<-OSS emulation
- ALSA: Clean up 64bit division function
ALSA<-OSS sequencer
- sound: seq_oss_midi: remove magic number
ARM AACI PL041 driver
- [ARM] 5544/1: Trust PrimeCell resource size
- [ARM] 5519/1: amba probe: pass "struct amba_id *" instead of void
ARM PXA2XX driver
- ASoC: Pass correct platform data from pxa2xx-ac9
- ALSA: Restore support for DMAless DAIs on PX
- ALSA: Allow passing platform_data for pxa2xx-ac9
- ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_fre
- pxa2xx-ac97: fix reset gpio mode settin
ATIIXP driver
- sound: Use PCI_VDEVIC
ATIIXP-modem driver
- sound: Use PCI_VDEVIC
AZT3328 driver
- ALSA: azt3328: fix previous breakage, improve suspend, cleanup
- ALSA: azt3328: large codec cleanup, add I2S port etc
Apple Onboard Audio driver
- ALSA: sound/aoa: Add kmalloc NULL test
- sound: remove driver_data direct access of struct devic
Au12x0/Au1550 PSC ASoC
- Add missing ASoC build stub
BT87x driver
- ALSA: bt87x - Add a quirk entry for Askey Computer Corp. MagicTView'9
- ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some driver
CA0106 driver
- ALSA: ca0106 - Fix the max capture buffer siz
- sound: Use PCI_VDEVICE for CREATIVE and ECTIV
- ALSA: ca0106 - Fix master volume scal
- ALSA: ca0106 - Add missing card->mixername field setu
- ALSA: Remove invalid GENERIC_MIX PCM sublas
- ALSA: ca0106 - Add missing registrations of vmaster control
- ALSA: Kill truncate warning by shortening Sigmatel-specific AC97 control nam
CMI8330 driver
- ALSA: cmi8330: Allow MPU-401-less operatio
- ALSA: cmi8330: find OPL3 port automaticall
- sound: cmi8330: Add basic CMI8329 suppor
- ALSA: cmi8330: revert comments about AD1848 bac
- ALSA: cmi8330: fix MPU-401 PnP init copy&paste bu
CMI8788 (Oxygen) driver
- sound: virtuoso: fix Xonar D1/DX silence after resum
- sound: oxygen: make mic volume control mon
- sound: virtuoso: add Xonar Essence ST suppor
- sound: virtuoso: enable HDAV S/PDIF inpu
- sound: virtuoso: add another DX PCI I
- sound: oxygen: reset DMA when stream is close
CMIPCI driver
- sound: Use PCI_VDEVIC
Conexant Riptide driver
- Regenerated riptide.patc
- ALSA: riptide - proper handling of pci_register_driver for joystic
- ALSA: riptide - Fix joystick resource handlin
- ALSA: riptide - Code clean u
- ALSA: riptide: postfix increment and off by on
Creative Sound Blaster X-Fi (20K1/20K2)
- Fix ctatc.patc
- Add missing pci/ctxfi/cttimer.
- ctxfi - Fix build with older kerne
- Add snd-ctxfi build stu
- ALSA: ctxfi - Simple code clean u
- ALSA: ctxfi - Fix uninitialized error check
- ALSA: ctxfi - Native timer support for emu20k
- ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k
- ALSA: ctxfi - Add PM suppor
- ALSA: ctxfi - Allow unknown PCI SSID
- ALSA: ctxfi - Fix deadlock with xfi-time
- ALSA: ctxfi - Replace atc lock to mute
- ALSA: ctxfi - Clear PCM resources at hw_params and hw_fre
- ALSA: ctxfi - Check the presence of SRC instance in PCM pointer callback
- ALSA: ctxfi - Add missing start check in atc_pcm_playback_start(
- ALSA: ctxfi - Add use_system_timer module optio
- ALSA: ctxfi - Fix wrong model id for UA
- ALSA: ctxfi - Clean up probe routine
- ALSA: ctxfi - Fix / clean up hw20k2 chip cod
- ALSA: ctxfi - Fix possible buffer pointer overru
- ALSA: ctxfi - Remove useless initializations and cas
- ALSA: ctxfi - Fix DMA mask for emu20k2 chi
- ALSA: ctxfi - Make volume controls more intuitiv
- ALSA: ctxfi - Optimize the native timer handling using wc counte
- ALSA: ctxfi - Add missing inclusion of linux/math64.
- ALSA: ctxfi - Set device 0 for mixer control element
- ALSA: ctxfi - Clean up / optimiz
- ALSA: ctxfi - Set periods_min to
- ALSA: ctxfi - Use native timer interrupt on emu20k
- ALSA: ctxfi - Fix previous fix for 64bit DM
- ALSA: ctxfi - Fix endian-dependent code
- ALSA: ctxfi - Allow 64bit DM
- ALSA: ctxfi - Support SG-buffer
- ALSA: ctxfi - Remove PAGE_SIZE limitatio
- ALSA: ctxfi - Fix supported PCM format
- ALSA: ctxfi - Fix PCM device namin
- ALSA: ctxfi - Fix surround mixer name
- ALSA: ALSA: ctxfi - Release PCM resources at each prepare cal
- ALSA: ctxfi - Fix Oops at mmappin
- ALSA: ctxfi - Fix a typo in MODULE_LICENS
- ALSA: ctxfi - Add missing module parameter definition
- ALSA: ctxfi - Move PCI ID definitions to linux/pci_ids.
- ALSA: ctxfi - Add missing inclusion of linux/delay.
- ALSA: ctxfi - Avoid unneeded pci_read_config_*() call
- ALSA: ctxfi - Add prefix to debug print
- ALSA: SB X-Fi driver merg
Digigram VX222 driver
- sound: vx222: fix input level control range chec
- trivial: fix typo milisecond/millisecond for documentation and source comments
Documentation
- ALSA: hda - Add / fix model entries for HD-audio drive
- ALSA: hda - Add quirk for MacBook Pro 5,5 with CS420
- ALSA: Add debug module optio
- ALSA: hda - Reword information messages for BIOS auto-probing mod
- ALSA: hda - Add description of new models for ALC889/889
- ALSA: pcm - Add logging of hwptr updates and interrupt update
- ALSA: hda - Merge patch_alc882() and patch_alc883(
- ALSA: hda - More description about patch module optio
- ALSA: hda - Add description about patch loadin
- ALSA: hda - Fix support for Samsung P50 with AD1986A code
- ALSA: hda - Add model=6530g optio
- trivial: Miscellaneous documentation typo fixe
- ALSA: pcm - Update document about xrun_debug proc fil
- ALSA: hda - Add 7.1 support for MSI GX62
- ALSA: support Sony Vaio T
- ALSA: ice1724 - Add ESI Maya44 suppor
- ALSA: hda - Acer Aspire 8930G suppor
- ALSA: Enable PCM hw_ptr_jiffies check only in xrun_debug mod
- ALSA: hda - Improved MacBook 3,1 suppor
- ALSA: SB X-Fi driver merg
- ALSA: hda - Add support of Samsung NC10 mini noteboo
- ALSA: hda - Add missing models for Realtek codec
- ALSA: sc6000: enable joystick por
- ALSA: hda - Addition for HP dv4-1222nr laptop suppor
- ASoC: Add power supply widget to DAP
- ALSA: Add missing description of lx6464es to ALSA-Configuration.tx
- ALSA: hda - Add 5stack-no-fp model for STAC927
- sound: virtuoso: add Xonar Essence ST suppor
EMU10K1/EMU10K2 driver
- Remove multiple KERN_ prefixes from printk format
- sound: Use PCI_VDEVICE for CREATIVE and ECTIV
- ALSA: emu10k1 - Fix minimum periods for efx playbac
- ALSA: Remove invalid GENERIC_MIX PCM sublas
- ALSA: clean up the logic for building sequencer module
ENS1370/1+ driver
- sound: Use PCI_VDEVICE for CREATIVE and ECTIV
ES1688 driver
- ALSA: Add missing __devexit_p() marker
Echoaudio driver
- ALSA: indigo-express: add missing 64KHz flag
Emagic Audiowerk 2
- trivial: typo (en|dis|avail|remove)bale -> (en|dis|avail|remove)abl
GUS Extreme driver
- ALSA: Add missing __devexit_p() marker
GUS Library
- ALSA: sound/isa: convert nested spin_lock_irqsave to spin_loc
Generic drivers
- time: move PIT_TICK_RATE to linux/timex.
- ALSA: pcsp - fix printk format warning agai
- ALSA: pcsp: fix printk format warnin
HDA Codec driver
- Add build stub for pci/hda/patch_cirrus.
- ALSA: hda - Fix probe of Toshiba laptops with ALC268 code
- ALSA: hda - Enable HP output with Macbook Pro 5,
- ALSA: hda - don't build digital output controls if not exis
- ALSA: hda - Fix compile warnings in patch_cirrus.
- ALSA: hda - Fix the speaker volume control nam
- ALSA: hda - Add GPIO setup for MacBook pro 5,5 with CS420
- ALSA: hda - Add quirk for MacBook Pro 5,5 with CS420
- ALSA: hda - Fix double creation of SPDIF input control
- ALSA: hda - Add CS420x-specific coef setu
- ALSA: hda - Force to initialize input mixer setup for CS420
- ALSA: hda - Fix cirrus codec parsin
- ALSA: hda - Add more quirk for HP laptops with AD1984
- ALSA: hda - Add full audio support on Acer Aspire 7730G noteboo
- ALSA: hda - Improve auto-cfg mixer name for ALC66
- ALSA: hda - Improve auto-cfg mixer name for ALC861-V
- ALSA: hda - Improve auto-cfg mixer name for ALC26
- ALSA: hda - Improve auto-cfg mixer name for ALC26
- ALSA: hda - Improve auto-cfg mixer name for ALC88
- ALSA: hda - Generalize input pin parsing in patch_realtek.
- ALSA: hda - Reuse ALC268 parser for ALC26
- ALSA: hda: move open coded tricks into get_wcaps_channels(
- ALSA: hda - Fix invalid capture mixers with some ALC268 model
- ALSA: hda - Add missing num_adc_nids definition for IDT92HD8xx
- ALSA: hda - Fix / clean up IDT92HD83xxx codec parse
- ALSA: hda - Enable line-out detection only with speaker
- ALSA: hda - fix noise issue when recording from digital mic with alc26
- ALSA: hda - Clean up init and setup hooks for Realtek codec
- ALSA: hda - Add setup hook to ALC preset struc
- ALSA: hda - Check connectivity for auto-mic of Realtek codec
- ALSA: hda - Use only one capture stream for auto-mi
- ALSA: hda - Add auto-mic support for Realtek codec
- ALSA: hda - Fix Oops due to STAC/IDT auto-mic change
- ALSA: hda - Add quirks for some HP laptop
- ALSA: hda - Fix line-out jack handling with STAC/IDT code
- ALSA: hda - Fix line-out jack detectio
- ALSA: hda: add IbexPeak/Clarkdale HDMI model with static cvt/pin numbe
- ALSA: hda - Add line-out jack detection on IDT/STAC codec
- ALSA: hda - Integrate Digital Input Source to Input Sourc
- ALSA: hda - Add Cirrus Logic CS420x suppor
- ALSA: hda: add model for Intel DG45ID/DG45FC board
- ALSA: hda: enable speaker output for Compaq 6530s/6531
- ALSA: hda - Don't override ADC definitions for ALC codec
- ALSA: hda - Add missing vmaster initialization for ALC26
- ALSA: hda - Read buffer overflo
- ALSA: hda: Correct EAPD for Dell Inspiron 152
- ALSA: hda: track CIRB/CORB command/response states for each code
- ALSA: hda - Fix quirk for Toshiba Satellite A135-S452
- ALSA: hda - Increase PCM stream name buf in patch_realtek.
- ALSA: hda - Fix typos of Capture controls
- ALSA: hda: add HP automute support to Intel ALC889/ALC889A model
- ALSA: hda: add 2-channel mode to Intel ALC889/ALC889A model
- ALSA: hda - No analog mix input source as default for IDT92HD71bx
- ALSA: hda - Add missing DMUX initialization for auto-mic with STAC/ID
- ALSA: hda - Remove static connection in IDT 92HD71bx
- ALSA: hda - Support auto-mic switching with IDT/STAC code
- ALSA: hda - Avoid overwrite of jack events with STAC/ID
- ALSA: hda - Don't create analog mixer for IDT92HD71bx
- ALSA: hda - Create Capture controls dynamicall
- ALSA: hda - Don't create unneeded digital input source for IDT 92HD71
- ALSA: hda - Reword information messages for BIOS auto-probing mod
- ALSA: hda - Add quirk for Dell Studio 155
- ALSA: hda - Add exception for volume-knob in snd_hda_get_connections(
- ALSA: hda - Introduce get_wcaps_type() macr
- ALSA: hda - Fix mute control with some ALC262 model
- [ALSA] Add better Intel IbexPeak platform suppor
- ALSA: hda - Restore GPIO1 properly at resume with AD1984
- ALSA: hda - Use snprintf() to be safe
- ALSA: hda - Fix ALC861 auto-mode parse
- ALSA: hda - Reduce click noise at power-savin
- ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codec
- ALSA: hda - Add quirk for Gateway T6834c lapto
- [ALSA] hda-intel: Cleanups for widget connection list handlin
- [ALSA] hda_codec: Check for invalid zero connection
- ALSA: hda - Fix ALC268 parser for mono speake
- ALSA: hda - Fix the previous sanity check in make_codec_cmd(
- ALSA: hda - add bounds checking for the codec command field
- ALSA: hda - Add CX20582 and OLPC XO-1.5 suppor
- ALSA: hda - Check codec errors in snd_hda_get_connections(
- ALSA: hda - Fix the merge erro
- ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checkin
- ALSA: hda - targa and targa-2ch fi
- ALSA: hda - Missing volume controls for Intel HDA (ALC269/EeePC
- ALSA: hda - Add quirks for RTL888 & RV630/M76 based MSI GX71
- ALSA: hda - Check widget types while parsing capture source in patch_via.
- ALSA: hda - Fix capture source selection in patch_via.
- ALSA: hda - Add missing EAPD initialization for VIA codec
- ALSA: hda - Clean up VT170x dig-in initialization cod
- ALSA: hda - move 8086:fb30 quirk (stac9205) to the proper sectio
- ALSA: hda - Don't override maxbps for FLOAT sharing with linear format
- ALSA: hda - Manually expand alc882_init_verb
- ALSA: hda - Add missing mixer amp initialization for ALC88
- ALSA: hda - Allow FLOAT PCM forma
- ALSA: hda - Fix input pinctl for ALC882 auto mod
- ALSA: hda - Merge patch_alc882() and patch_alc883(
- ALSA: hda - Add patch module optio
- ALSA: hda - Don't call snd_hda_codec_configure in snd_hda_codec_new(
- ALSA: hda - Avoid invalid formats and rates with shared SPDI
- ALSA: hda - Improve ASUS eeePC 1000 mixe
- ALSA: hda - Add GPIO1 control at muting with HP laptop
- ALSA: hda - Add quirk for HP 6930
- ALSA: hda - Add missing static to patch_ca0110(
- ALSA: hda - Add missing initializations for ALC268 and ALC26
- ALSA: hda - Line In for Acer Inspire 6530G mode
- ALSA: hda - Use model=acer-aspire-6530g for Acer Aspire 6930
- ALSA: hda - Fix acer-aspire-6530g model quir
- ALSA: hda - Add pin-sense trigger when needed for Realtek codec
- ALSA: hda - Fix support for Samsung P50 with AD1986A code
- ALSA: hda - Generalize the pin-detect quirk for Lenovo N10
- ALSA: hda - Simplify AD1986A mixer definition
- ALSA: hda - Make jack-plug notification selectabl
- ALSA: hda - Add digital-mic support to ALC262 auto mode
- ALSA: hda - Fix check of input source type for realtek codec
- ALSA: hda - Add quirk for Sony VAIO Z21M
- ALSA: hda - Get back Input Source for ALC262 toshiba-s06 mode
- ALSA: hda - Fix unsigned comparison in patch_sigmatel.
- ALSA: hda - Add model=6530g optio
- ALSA: hda - Acer Inspire 6530G model for Realtek ALC88
- ALSA: HDA - Correct trivial typos in comments
- ALSA: HDA - Name-fixes in code (tagra/targa
- ALSA: HDA - Add pci-quirk for MSI MS-7350 motherboard
- ALSA: hda - Fix memory leak at codec creatio
- ALSA: hda - Add quirk for Acer Aspire 6935
- ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205
- ALSA: hda - Fix the previous tagra-8ch patc
- ALSA: hda - Add 7.1 support for MSI GX62
- ALSA: support Sony Vaio T
- ALSA: hda - More Aspire 8930G fixe
- ALSA: hda - Limit codec-verb retry to limited hardware
- ALSA: hda - Add codec bus reset and verb-retry at critical error
- ALSA: hda - Acer Aspire 8930G suppor
- ALSA: hda - Reorder and clean-up ALC268 quirk tabl
- ALSA: hda - fix audio on LG R51
- ALSA: hda - Macbook[Pro] 5 6ch suppor
- ALSA: hda - Jack Mode changes for Sigmatel board
- ALSA: hda - Support NVIDIA 8 channel HDMI audi
- ALSA: hda-intel: improve initialization for ALC262_HP_BPC mode
- ALSA: hda - Fix reverted LED setup for H
- ALSA: hda - Use snd_hda_codec_get_pincfg() in patch_ca0110.
- ALSA: hda - Fix channels_max setting for CA011
- ALSA: hda - Minor clean up of patch_sigmatel.
- ALSA: hda - Compaq Presario CQ60 patching for Conexan
- ALSA: hda - Support sync after writing a ver
- ALSA: hda - Fix digital beep tone calculatio
- ALSA: hda - Improved MacBook 3,1 suppor
- ALSA: hda - Show the actual chip name in 'unkown model' message
- ALSA: hda - Split codec->name to vendor and chip name string
- ALSA: hda - add controls to toggle DC bias on mic port
- ALSA: hda - Add a quirk entry for Macbook Pro 5,
- ALSA: hda - Disable fallback to model=acer for Acer laptop
- ALSA: hda - Add support of Samsung NC10 mini noteboo
- ALSA: hda - Add missing models for Realtek codec
- ALSA: hda - Clean up Realtek auto-mute unsol routine
- ALSA: hda - Clean up for ALC262 HP model auto-mute function
- ALSA: hda - Fix and clean up hippo-compat HP auto-mutin
- ALSA: hda - Fix secondary SPDIF on VT1708S and VT1702 codec
- ALSA: hda - Add support for MacBook 5.1 (Aluminium
- ALSA: hda - Addition for HP dv4-1222nr laptop suppor
- ALSA: hda - Fix a typo in patch_realtek.c agai
- ALSA: hda - Don't enable auto-mute but for speakers in patch_realtek.
- ALSA: hda - Add amp initialization for realtek auto mod
- ALSA: hda - Fix a typo in debug print for realtek auto-detectio
- ALSA: hda - minor optimization in hda_set_power_state(
- ALSA: hda - Add debug prints for Realtek auto-ini
- ALSA: hda - Retry codec-verbs at error
- ALSA: hda - Cache PCM and STREAM parameters querie
- ALSA: hda - Check strcpy lengt
- ALSA: hda - Add Creative CA0110-IBG suppor
- ALSA: hda - Add missing check of pin vref 50 and others in Realtek codec
- ALSA: hda - Add 5stack-no-fp model for STAC927
- ALSA: hda - fix audio on HP TX25xx series notebook
- ALSA: hda - Fix line-in on Mac Mini Core2 Du
HDA Intel driver
- Fix build of hda_intel.
- ALSA: hda - Add a white-list for MSI optio
- ALSA: hda: warn on spurious respons
- ALSA: hda: remember last command for each code
- ALSA: hda: read CORBWP inside reg_loc
- ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_i
- ALSA: hda: take cmd_mutex in probe_codec(
- ALSA: hda: track CIRB/CORB command/response states for each code
- ALSA: hda - Add support for new AMD HD audio device
- ALSA: hda - Disable AMD SB600 64bit address support onl
- ALSA: hda - Fix error path in the sanity check in azx_pcm_open(
- ALSA: hda - Add patch module optio
- ALSA: hda - Don't call snd_hda_codec_configure in snd_hda_codec_new(
- ALSA: hda - Add sanity check in PCM open callbac
- ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callbac
- ALSA: hda_intel: fix build error when !P
- ALSA: hda - Limit codec-verb retry to limited hardware
- ALSA: hda - Add codec bus reset and verb-retry at critical error
- ALSA: hda - Fix a typo in the previous patc
- ALSA: hda - Add more register bits definition
- ALSA: hda - Always sync writes in single_cmd mod
- ALSA: hda - Allow concurrent RIRB access in single_cmd mod
- ALSA: hda - Reset CORB/RIRB at retrying the verb communicatio
- ALSA: hda - Add prefix to kernel message
- ALSA: hda - Avoid conflicts with snd-ctxfi drive
- ALSA: hda - Retry codec-verbs at error
- ALSA: hda - Check strcpy lengt
- ALSA: hda - Add Creative CA0110-IBG suppor
- ALSA: hda - Add forced codec-slots for ASUS W5F
HDA generic driver
- Fix build of hda_intel.
- ALSA: hda: move open coded tricks into get_wcaps_channels(
- ALSA: hda - Add Cirrus Logic CS420x suppor
- ALSA: hda: fix out-of-bound hdmi_eld.sad[] writ
- ALSA: hda - Introduce get_wcaps_type() macr
- [ALSA] hda_generic: use AC_WCAP_CONN_LIST check for widget connection
- [ALSA] hda_generic: do not read connections for widged with an unknown typ
- ALSA: hda - fix beep tone calculation for IDT/STAC codec
- ALSA: hda - Check "beep" hin
- ALSA: hda - Add patch module optio
- ALSA: hda - Don't call snd_hda_codec_configure in snd_hda_codec_new(
- ALSA: hda - Make jack-plug notification selectabl
- ALSA: hda - Fix digital beep tone calculatio
- ALSA: hda - Split codec->name to vendor and chip name string
- ALSA: hda - Add Creative CA0110-IBG suppor
I2C UDA1380
- ASoC: UDA1380: refactor device registratio
ICE1712 driver
- Add build stub for ice1724 maya44 suppor
- ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD
- ALSA: ice1724 - Add ESI Maya44 suppor
- ALSA: ice1724 - Allow spec driver to create own routing control
ICE1724 driver
- ALSA: ice1724 - Fix section mismatc
- ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD
- ALSA: ice1724 - Add ESI Maya44 suppor
- ALSA: ice1724 - Allow spec driver to create own routing control
- ALSA: ice1724 - Add PCI postint to reset sequenc
- ALSA: ice1724 - Clean up definitions of DMA record
- ALSA: ice1724 - Check error in set_rate functio
ISA
- ALSA: sc6000: add support for SC-6600 and SC-700
Intel8x0 driver
- ALSA: intel8x0 - Fix PCM position crazines
KORG1212 driver
- ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some driver
LX6464ES
- ALSA: lx6464es - configure ethersound io channel
- convert some DMA_nnBIT_MASK() caller
- ALSA: lx6464es - support standard alsa module parameter
- ALSA: lx6464es - Disable lx_message_send(
- ALSA: lx6464es - Use snd_card_create(
- ALSA: lx6464es - driver for the digigram lx6464es interfac
MSND driver
- ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some driver
Memalloc module
- ALSA: Fix SG-buffer DMA with non-coherent architecture
OPL3
- ALSA: clean up the logic for building sequencer module
OPL4
- ALSA: clean up the logic for building sequencer module
OSS device core
- sound: make OSS device number claiming optional and schedule its remova
- sound: request char-major-* module aliases for missing OSS device
- sound: do not set DEVNAME for OSS device
- Driver Core: sound: add nodename for sound driver
PARISC Harmony driver
- ALSA: Add missing __devexit_p() marker
- ALSA: parisc/harmony: fix printk format warnin
PCI drivers
- ALSA: azt3328: fix Kconfig entr
- ALSA: ctxfi - Remove PAGE_SIZE limitatio
- ALSA: ctxfi - Add depends on X8
- ALSA: SB X-Fi driver merg
- ALSA: hdsp - Add a comment about external firmwares for hds
- ALSA: lx6464es - driver for the digigram lx6464es interfac
- sound: virtuoso: add Xonar Essence ST suppor
PDAudioCF driver
- ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some driver
PPC AWACS driver
- ALSA: powermac - Replace the rest of __init
- ALSA: sound/ppc: update annotations of serveral function
PPC Beep
- ALSA: sound/ppc: update annotations of serveral function
PPC Burgundy driver
- ALSA: burgundy: timeout message is off by one
- ALSA: powermac - Replace the rest of __init
- ALSA: sound/ppc: update annotations of serveral function
PPC DACA driver
- ALSA: sound/ppc: update annotations of serveral function
PPC Keywest driver
- ALSA: keywest: Get rid of useless i2c_device_name() macr
PPC PMAC driver
- ALSA: powermac - Replace the rest of __init
PPC PS3 driver
- ALSA: sound/ps3: Correct existing and add missing annotation
- ALSA: sound/ps3: Restructure driver sourc
- ALSA: sound/ps3: Fix checkpatch issue
PPC Tumbler driver
- ALSA: powermac - Replace the rest of __init
RME HDSP driver
- ALSA: hdsp - allow proc reporting with disconnected io bo
- ALSA: Clean up 64bit division function
- ALSA: hdsp: allow firmware loading from inside the kerne
RME9652 driver
- ALSA: Clean up 64bit division function
SB drivers
- ALSA: clean up the logic for building sequencer module
SC6000 (CompuMedia ASC-9308 + AD1848) driver
- ALSA: sc6000: enable joystick por
- ALSA: sc6000: fix older card initializatio
- ALSA: sc6000: add support for SC-6600 and SC-700
SGI O2 Audio
- ALSA: sgio2audio.c: clean up checkin
SIS7019 driver
- trivial: fix typos s/paramter/parameter/ and s/excute/execute/ in documentation and source comments
SoC Audio for Freecale i.MX1x i.MX2x CPUs
- Add soc/imx/* build stu
- ASoC: Staticise unexported variable
- ASoC: Remove unneeded i.MX dependency on SN
- ASoC: Fix review issues in i.MX2x PCM drive
- ASoC: add machine driver for i.mx27_visstrim_m10 boar
- ASoC: add DAI platform ssi driver for MX
- ASoC: add DMA platform driver for MX1x and MX2
SoC Audio for TXx9
- Add soc/txx9 build stu
- ASoC: txx9aclc: dynamically allocate dmaengine devnam
- ASoC: Kill BUS_ID_SIZ
- ASoC: Add TXx9 AC link controller driver (v3
SoC Audio for the Atmel AT32/AT91 System-on-Chip
- Add missing ASoC build stub
- ASoC: Configure WM8731 SYSCLK at startup on AT91SAM9G20-E
- ASoC: Disable microphone input for AT91SAM9G20-EK by defaul
- ASoC: Use CODEC as clock master on AT91SAM9G20-E
- ASoC: correct print specifiers for unsigned
- ASoC: AFEB9260 drive
SoC Audio for the Samsung S3C24XX chips
- ASoC: neo1973_gta02_wm8753: Replace deprecated s3c_gpio calls with gpioli
- ASoC: neo1973_gta02_wm8753: Replace snd_soc_cnew with snd_soc_add_controls
- ASoC: Fix s3c-i2s-v2 buil
- ASoC: Add S3C24xx dependencies for Simtec machine
- ASoC: S3C platform: Fix s3c2410_dma_started() called at improper tim
- ASoC: Select core DMA when building for S3C64x
- ASoC: S3C24XX: Support for Simtec Hermes board
- ASoC: S3C24XX: Add audio core and tlv320aic23 for Simtec board
- ASoC: S3C24XX : Align the peroid size to the buffer siz
- ASoC: Reenable S3C64xx I2S suppor
- ASoC: Fix data format configuration for S3C64XX IISv
- ASoC: s3c2443-ac97: convert semaphore to mute
- ASoC: Existing S3C24xx AC97 drivers should depend on S3C24x
- ASoC: Add Openmoko Neo FreeRunner (GTA02) audio drive
- ASoC: Fix lm4857 contro
- [ARM] S3C24XX: GPIO: Move gpio functions out of <mach/hardware.h
- [ARM] S3C24XX: Remove hardware specific registers from DM
- ASoC: Use platform device resource for S3C64xx IISv
- ASoC: Staticise txctrl and rxctrl for S3C IISv
- ASoC: Display S3C IISv2 mode and MS errors by defaul
- ASoC: Display the clock rate used as the basis for rate calculatio
- ASoC: Allow use of resource from the platform device for S3C IISv
- ASoC: Fix boot warnings from S3C IISv
- ASoC: Fix data format configuration for S3C64xx IISv2 and add 24 bi
- ASoC: Make S3C64xx clock export function to return struct cl
- ASoC: Check for supported CPUs when building s3c-i2s-v
- ASoC: Fix error message formatting in s3c64xx-i2s drive
- ASoC: Use our registration function for S3C64x
- ASoC: s3c-i2s-v2 diagnostic improvement
- ASoC: Enforce symmetric rates for S3C64xx I2S interfac
- ASoC: S3C2412: Failing to get the I2S clock is an erro
- ASoC: Fix S3C64xx IIS device registration and support both port
SoC Blackfin
- ASoC: Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm da
- ASoC: board driver to connect bf5xx with ad193
- ASoC: blackfin I2S(TDM mode) CPU DAI drive
- ASoC: Blackfin I2S: fix resume handlin
- ASoC: Blackfin AC97: fix resume handlin
- ASoC: Blackfin: convert internal names from bf52x to bf5x
- ASoC: Blackfin: update the bf5xx_i2s_resume parameter
- ASoC: Blackfin: keep better track of SPORT configuration stat
- ASoC: Blackfin: document how anomaly 05000250 is handle
- ASoC: Blackfin: set the transfer size according the ac97_frame siz
SoC Codec AC97
- ASoC: Use a shared define for AC97 CODEC data format
SoC Codec AD1836
- Add more missing build stubs for ASo
- ASoC: Minor cleanups to AD1938 drive
- sound: new ad1836 codec driver based on aso
SoC Codec AD1938
- ASoC: delete -spi suffix in ad1938 and free private data while registers fai
- ASoC: add output/input widgets in ad1938 to make dac/adc dynamic PM wor
- ASoC: Update AD1938 for new TDM slot AP
- ASoC: Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm da
- ASoC: Fix checkpatch issues in AD193
- ASoC: Kill direct accesses to driver_dat
- ASoC: new ad1938 codec driver based on aso
SoC Codec AD1980
- ASoC: Use a shared define for AC97 CODEC data format
SoC Codec AK4535
- ASoC: Remove unused AK4535 hardware read functionalit
SoC Codec AK4642
- ASoC: Add ak4642/ak4643 codec suppor
SoC Codec CS4270
- ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_devic
- ASoC: cs4270: add power management suppor
- ASoC: cs4270: introduce CS4270_I2C_INC
- ASoC: cs4270: add Master Playback Switc
- ASoC: cs4270: fix Master Capture Switch polarit
SoC Codec CX20442
- ASoC: CX20442: simplify codec controller usag
- ASoC: CX20442: add some debuggin
- ASoC: CX20442: push down machine independent line discipline bit
- ASoC: CX20442: fix issues pointed out by subsystem maintaine
- ASoC: Add support for Conexant CX20442-11 voice modem code
SoC Codec DIT SPDI/F
- ASoC: spdif: set module licence to GP
- ASoC: spdif codec: enable use by module
- ASoC: Initialise dev for the dummy S/PDIF DA
- ASoC: Add dummy S/PDIF codec suppor
SoC Codec MAX9877
- ASoC: MAX9877: fix write operation for registe
- ASoC: MAX9877: separate callback function
- ASoC: MAX9877: add MAX9877 amp drive
SoC Codec Philips UDA134x
- ASoC: UDA134X: Fix mistaken mute/unmute cod
SoC Codec Philips UDA1380
- ASoC: UDA1380: refactor device registratio
SoC Codec SSM2602
- ASoC: Revert duplicated code in SSM2602 drive
- ASoC: SSM2602: assign last substream to the master when shutting dow
- ASoC: SSM2602: remove unsupported sample rate
SoC Codec STAC9766
- ASoC: Keep index within stac9766_reg[
- ASoC: Fix minor issues in STAC9766 drive
- ASoC: Codec for STAC9766 used on the Efik
SoC Codec TLV320AIC23
- ASoC: codec tlv320aic23 fix bogus divide by 0 messag
- ASoC: correct print specifiers for unsigned
- ASoC: tlv320aic23: add DSP_A format suppor
SoC Codec TLV320AIC3X
- ASoC: Make platform data optional for TLV320AIC3
- ASoC: tlv320aic3x: Change to use device mode
- ASoC: Remove use of hw_read from TLV320AIC3x drive
- ASoC: tlv320aic3x: Enable PLL when not bypasse
SoC Codec TWL4030
- ASoC: TWL4030: Fix for capture mixer string
- ASoC: TWL4030: Introduce PGAs for output
- ASoC: TWL4030: Add tristate callbacks for HiFi and Voic
- ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effec
- ASoC: Remove word "Switch" from Handsfree switch nam
- ASoC: TWL4030: Correct bypass event for voice sideton
- ASoC: TWL4030: Add AVADC Clock Priorit
- ASoC: TWL4030: Fix voice interface clock master
- ASoC: Staticise put_twl4030_opmode_enum_double(
- ASoC: Fix shadowed variables in twl403
- ASoC: Fix build error in twl4030.
- ASoC: TWL4030: Check the interface format for 4 channel mod
- ASoC: TWL4030: Use reg_cache in twl4030_init_chi
- ASoC: TWL4030: HandsfreeL/R mute DAPM switc
- ASoC: TWL4030: Add shadow registe
- ASoC: TWL4030: Handsfree pop removal redesig
- ASoC: TWL4030: Differentiate the playback stream
- ASoC: TWL4030: Add support for platform dependent configuratio
- ASoC: TWL4030: Move the Headset pop-attenuation code to PGA even
- ASoC: TWL4030: Change DAPM routings and controls for DACs and PGA
- ASoC: TWL4030: Add control for selecting codec operation mod
- ASoC: TWL4030: Fix Analog capture path for AUX
- ASoC: TWL4030: Enable/disable voice digital filter
- ASoC: TWL4030: change DAPM for analog microphone selectio
- ASoC: TWL4030: Fix typo in twl4030_codec_mute functio
- ASoC: TWL4030: Add VIBRA outpu
- ASoC: TWL4030: Add voice digital loopback: sideton
- ASoC: TWL4030: Add VDL analog bypas
- ASoC: TWL4030: Add 4 channel TDM suppor
- ASoC: TWL4030: Add VDL path suppor
- ASoC: TWL4030: Add support Voice DA
- ASoC: TWL4030: Fix for the constraint handlin
- ASoC: TWL4030: Fix gain control for earpiece amplifie
SoC Codec WM8350
- ASoC: Don't reconfigure WM8350 FLL if not neede
- ASoC: Add suspend and resume callbacks to Wolfson CODEC driver
- ASoC: Automatically manage WM8350 sloping stopband filte
- ASoC: Include WM8350 register definitions in CODEC heade
- ASoC: Fix logic in WM8350 master clocking chec
SoC Codec WM8400
- ASoC: Bodge around GCC 4.4.0 flow analysis bug in GCC 4.4.
- ASoC: Add suspend and resume callbacks to Wolfson CODEC driver
- ASoC: remove driver_data direct access of struct devic
SoC Codec WM8510
- ASoC: Factor out 7 bit register 9 bit data SPI writ
- ASoC: Add I/O control bus information to factored out cache setu
- ASoC: Begin to factor out register cache I/O function
- ASoC: WM8510 has a single frame clock so needs symmetric rate
SoC Codec WM8523
- ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODEC
- ASoC: Add WM8523 CODEC drive
SoC Codec WM8580
- ASoC: Add I/O control bus information to factored out cache setu
- ASoC: Factor out WM8580 register cache cod
- ASoC: Regulator support for WM858
- ASoC: Add suspend and resume callbacks to Wolfson CODEC driver
SoC Codec WM8728
- ASoC: Factor out 7 bit register 9 bit data SPI writ
- ASoC: Add I/O control bus information to factored out cache setu
- ASoC: Begin to factor out register cache I/O function
SoC Codec WM8731
- ASoC: Drop unneeded declaration of removed wm8731 SPI write functio
- ASoC: Factor out 7 bit register 9 bit data SPI writ
- ASoC: Limit WM8731 to symmetric rate
- ASoC: Correct WM8731 Mic Capture Switch control nam
- ASoC: Add TLV information for WM873
- ASoC: Fix leaks in WM8731 probe error handlin
- ASoC: Add suspend and resume callbacks to Wolfson CODEC driver
- ASoC: remove driver_data direct access of struct devic
SoC Codec WM8750
- ASoC: Factor out 7 bit register 9 bit data SPI writ
SoC Codec WM8753
- ASoC: Fix wm8753 register cache size and initializatio
- ASoC: Fix register cache initialisation for WM875
- ASoC: remove driver_data direct access of struct devic
SoC Codec WM8776
- ASoC: Convert WM8776 to use factored out register cache cod
- ASoC: Add WM8776 CODEC drive
SoC Codec WM8900
- ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODEC
- ASoC: Automatically manage WM8900 sloping stopband filte
SoC Codec WM8903
- ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODEC
- ASoC: Automatically control WM8903 sloping stopband filte
- ASoC: Remove odd bit clock ratios for WM890
- ASoC: Implement WM8903 digital sidetone suppor
- ASoC: Remove redundant rate constraint for WM890
- ASoC: Actively manage the DC servo for WM890
- ASoC: Optimise configuration of WM8903 DC serv
- ASoC: Support CLK_DSP in WM890
- ASoC: Use DAPM supply widget for WM8903 charge pum
- ASoC: Request shared rates for WM890
SoC Codec WM8940
- ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODEC
- ASoC: Add missing __devexit in wm8940.
- ASoC: Staticise TLV values in WM894
- sound: ASoC WM8940 Drive
SoC Codec WM8960
- ASoC: Fix WM8960 leaks on probe failur
- ASoC: Add WM8960 CODEC drive
SoC Codec WM8961
- ASoC: Fix WM8961 suspend function typ
- ASoC: Add core suspend and resume callbacks to WM896
- ASoC: Add WM8961 drive
SoC Codec WM8974
- Add more missing build stubs for ASo
- ASoC: Factor out cache I/O from WM897
- ASoC: Correct a bug with "ADC Inversion Switch" in wm8974 codec
- ASoC: WM8974 DAPM cleanup
- ASoC: WM8974 cosmetic cleanup
- ASoC: Use symmetric rates for WM897
- ASoC: Add WM8974 TLV informatio
- ASoC: Refresh WM8974 PLL configuratio
- ASoC: Declare 2 channels for WM897
- ASoC: Refresh WM8974 bias configuratio
- ASoC: Remove unreferenced wm8974_add_controls(
- ASoC: Update WM8974 to use standard I2C device probe method
- ASoC: WM8974 checkpatch cleanup
- ASoC: Add WM8974 CODEC drive
SoC Codec WM8988
- Sound: remove direct access of driver_dat
- ASoC: Fix leaks in WM8988 registration error handlin
- ASoC: Add WM8988 CODEC drive
SoC Codec WM8990
- ASoC: Fix errors in WM899
SoC Codec WM8993/4
- Add more missing build stubs for ASo
- ASoC: Remove unneeded inclusion of linux/regulator/consumer.
- ASoC: Remove duplicate ADC/DAC widgets from wm_hubs.
- ASoC: WM8993 digital mixing suppor
- ASoC: Implement TDM configuration for WM899
- ASoC: Fix WM8993 MCLK configuration for high frequency MCLK
- ASoC: Factor out shared code from WM899
- ASoC: Fix FLL reference clock division setup in WM899
- ASoC: Fix sample rate lookup in WM899
- ASoC: Add WM8993 CODEC drive
SoC Codec WM9081
- ASoC: Update WM9081 for tdm_slot() API chang
- ASoC: change set_tdm_slot api to allow slot_width override
- ASoC: Error out if we can't determine a suitable WM9081 syscl
- ASoC: Fix WM9081 PowerPC compiler issue
- ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier drive
SoC Codec WM9705
- ASoC: free socdev if init_card() fails in wm9705_soc_probe(
- ASoC: Use a shared define for AC97 CODEC data format
SoC Codec WM9712
- ASoC: Support AC97 link off by default on WM971
SoC Codec WM9713
- ASoC: Move the WM9713 voice DAC powerdown to a DAPM even
- ASoC: WM9713 requires symmetric rates on the voice DA
SoC DaVinci
- ASoC: tlv320aic3x: fixup board device change
- ASoC: tlv320aic3x: Change to use device mode
- ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EV
- ASoC: DaVinci: Add a DAI format to McASP drive
- ASoC: DaVinci: McASP driver enhacement
- ASoC: DaVinci: Support Audio on DA830 EV
- ASoC: DaVinci: pcm, constrain buffer size to multiple of perio
- ASoC: DaVinci: i2s: don't bounce through rtd to get da
- ASoC: davinci: don't use clock name
- ASoC: Introduce platform driver model for dm644x, dm35
- ASoC: DaVinci I2S needs mach/asp.
- ASoC: DaVinci: pcm, don't play 1st sound period twic
- ASoC: Add machine driver support for DM646
- ASoC: Add mcasp support for DM646
- ASoC: DaVinci: i2s, add davinci_i2s_prepare and shutdow
- ASoC: DaVinci: i2s, fix mcbsp_word_length updat
- ASoC: DaVinci: i2s, minor cleanup of davinci_i2s_startu
- ASoC: DaVinci: i2s, only start sample generator if neede
- ASoC: DaVinci: i2s cleanu
- ASoc: DaVinci: i2s, minor cleanu
- ASoC: DaVinci: i2s toggle clock to complete rese
- ASoC: DaVinci: i2s, remove MOD_REG_BIT macr
- ASoC: DaVinci EVM board support buildfixe
- ASoC: DaVinci I2S update
- ASoC: davinci-pcm buildfixe
SoC Dynamic Audio Power Management
- ASoC: add missing inclusion of debugfs.
- ASoC: Add DAPM widget power decision debugfs file
- ASoC: Provide default set_bias_level() implementatio
- ASoC: Add input and output AIF widget
- ASoC: Power speakers and headphones simultaneousl
- ASoC: Fix handling of bias levels for non-DAPM codec
- ASoC: fix checking for external widgets bu
- ASoC: Add pop delay debug at end of DAPM sequencin
- ASoC: Fix widget powerdown on shutdow
- ASoC: Add a shutdown callbac
- ASoC: Make DAPM power sequence lists local variable
- ASoC: Coalesce power updates for PGA
- ASoC: Coalesce power updates for DAPM widgets with event
- ASoC: Sort specialised mixers and muxes togethe
- ASoC: Coalesce register writes for DAPM sequence
- ASoC: Allow 32 bit registers for DAP
- ASoC: Factor out DAPM sequence executio
- ASoC: Sort DAPM power sequences while building list
- ASoC: Apostrophe patro
- ASoC: Add debug trace for bias level transition
- ASoC: Integrate bias management with DAPM power managemen
- ASoC: Make DAPM sysfs entries non-optiona
- ASoC: Split DAPM power checks from sequencing of power change
- ASoC: Add power supply widget to DAP
- ASoC: Make the DAPM power check an operation on the widge
- ASoC: Factor out DAPM power checks for DACs and ADC
- ASoC: Factor out generic widget power check
- ASoC: Support DAPM events for DACs and ADC
- ASoC: Factor out application of power for generic widget
- ASoC: Display return code when failing to add a DAPM kcontro
SoC FSI SH7724
- ASoC: Add SuperH FSI driver support for ALS
SoC Freescale
- ASoC: MPC5200: Support for buffer wrap aroun
- ASoC: Add missing DRV_NAME definitions for fsl/* driver
- ASoC: MPC5200: Increase the delay time between reset
- ASoC: add locking to mpc5200-psc-ac97 drive
- ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleare
- ASoC: remove BROKEN from Efika and pcm030 fabric driver
- ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfi
- ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout(
- ASoC: Switch FSL SSI DAI over to symmetric_rate
- ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolve
- ASoC: Fabric bindings for STAC9766 on the Efik
- ASoC: Support for AC97 on Phytec pmc030 base board
- ASoC: AC97 driver for mpc520
- ASoC: Main rewite of the mpc5200 audio DMA cod
- ASoC: Rename the PSC functions to DM
- ASoC: Basic split of mpc5200 DMA code out of mpc5200_psc_i2
- sound: use dev_set_drvdat
- ASoC: Remove BROKEN from mpc5200 kconfi
- ASoC: Set the MPC5200 i2s driver to BROKEN status
SoC Layer
- Fix build of soc-core.c with older kernel
- ASoC: fix I2C build error
- ASoC: Add DAPM widget power decision debugfs file
- ASoC: Add ak4642/ak4643 codec suppor
- ASoC: Hook i.MX into buil
- ASoC: Factor out shared code from WM899
- ASoC: Minor cleanups to AD1938 drive
- sound: new ad1836 codec driver based on aso
- ASoC: Define more formats for the AC97 CODEC
- ASoC: change set_tdm_slot api to allow slot_width override
- ASoC: Add WM8776 CODEC drive
- ASoC: Factor out I2C 8 bit address 16 bit data I/
- ASoC: Add I/O control bus information to factored out cache setu
- ASoC: jack: Fix race in snd_soc_jack_add_gpio
- ASoC: Allow CODECs to flag invalid register
- ASoC: Begin to factor out register cache I/O function
- ASoC: Add WM8974 CODEC drive
- ASoC: Jack handling enhancements as suggested by subsystem maintaine
- ALSA: Allow passing platform_data to devices attached to AC97 bu
- ASoC: Add support for Conexant CX20442-11 voice modem code
- ASoC: new ad1938 codec driver based on aso
- ASoC: MAX9877: add MAX9877 amp drive
- ASoC: add SOC_DOUBLE_R_EXT_TLV control typ
- ASoC: add SOC_DOUBLE_EXT_TLV control typ
- ASoC: fixes multiple typos in comments, no functional chang
- ASoC: Add WM8993 CODEC drive
- ASoC: Add CODEC volatile register operatio
- ASoC: Add WM8523 CODEC drive
- ASoC: Convert to dev_pm_op
- ASoC: Add a shutdown callbac
- ASoC: Add stub suspend and resume calls for ASoC subdevice
- ASoC: Add WM8961 drive
- ASoC: Make DAPM power sequence lists local variable
- ASoC: Allow 32 bit registers for DAP
- ASoC: Instantiate any forgotten DAPM widget
- ASoC: fix NULL pointer dereference in soc_suspend(
- ASoC: Add dummy S/PDIF codec suppor
- ASoC: Codec for STAC9766 used on the Efik
- ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier drive
- AsoC: Make snd_soc_read() and snd_soc_write() function
- ASoC: Add TXx9 AC link controller driver (v3
- ASoC: Integrate bias management with DAPM power managemen
- ASoC: Split DAPM power checks from sequencing of power change
- ASoC: Add SNDRV_PCM_FMTBIT_S32_BE as a valid AC97 forma
- ASoC: Fix up CODEC DAI formats for big endian CPU
- ASoC: Remove redundant codec pointer from DAI
- ASoC: Remove unused DAI format define
- ASoC: Use a shared define for AC97 CODEC data format
- sound: ASoC WM8940 Drive
- ASoC: add SOC_DOUBLE_EXT macr
- ASoC: Volume controls are never of boolean typ
- ASoC: Check we have DAI ops when calling via accessor function
- ASoC: Add WM8960 CODEC drive
- ASoC: Add WM8988 CODEC drive
- ASoC: Provide core support for symmetric sample rate
- ASoC: soc-core: fix crash when removing not instantiated car
- ASoC: Add driver for s6000 I2S interfac
SoC PXA2xx Corgi
- [ARM] pxa: register wm8731 explicitly for corgi and poodl
SoC PXA2xx EM-X270
- ASoC: em-x270: make the driver support also eXeda and CM-X300 machine
SoC PXA2xx Palm T|X
- ASoC: Switch palm27x-asoc to jack detection ap
- [ARM] pxa/palm: Add Palm27x aSoC driver to PalmTE
SoC PXA2xx Poodle
- [ARM] pxa: register wm8731 explicitly for corgi and poodl
SoC S6000
- ASoC: tlv320aic3x: fixup board device change
- ASoC: tlv320aic3x: Change to use device mode
- ASoC: correct s6000 I2S clock polarit
- ASoC: s6105 IP camera machine specific ASoC cod
- ASoC: Add driver for s6000 I2S interfac
SoC SH7760 AC97
- ASoC: Add FSI-AK4642 sound support for Super
- ASoC: Add SuperH FSI driver support for ALS
SoC Texas Instruments OMAP
- sound: TTY/ASoC: Rename N_AMSDELTA line discipline to N_V25
- ASoC: SDP3430: Fix TWL GPIO6 pin mux reques
- sound: ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_sto
- ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DA
- ASoC: tlv320aic3x: fixup board device change
- ASoC: tlv320aic3x: Change to use device mode
- ASoC: OMAP: Use DMA operating mode of McBS
- ASoC: OMAP: Use McBSP threshold to playback and captur
- ASoC: Always syncronize audio transfers on frame
- ASoC: Add runtime check for RFIG and XFI
- ASoC: OMAP: Make DMA 64 aligne
- ASoC: OMAP: Enable DMA burst mod
- ASoC: OMAP: Enhance OMAP1510 DMA progress software counte
- ASoC: OMAP: Make use of DMA channel self linking on OMAP151
- sound: ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/sto
- ASoC: add support for Amstrad E3 (Delta) machin
- ASoC: OMAP: Staticise pcm creation function of omap-pc
- ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO
- ASoC: Zoom2: Update twl4030_setup_data parameter
- ASoC: TWL4030: Fix voice interface clock master
- ASoC: Zoom2: Add machine driver for Zoom2 boar
- ASoC: OMAP: fix OMAP1510 broken PCM pointer callbac
- ASoC: SDP4030: Use the twl4030_setup_data for headset pop-remova
- ASoC: SDP3430: Connect twl4030 voice DAI to McBSP
- ASoC: Added OMAP3 EVM support in ASoC
- ASoC: Beagle: Add support for 4 channe
- ASoC: OMAP: Add 4 channel support to mcbs
- ASoC: OMAP: Add checking to detect bufferless pcm
- ASoC: TWL4030: Add support Voice DA
- ASoC: OMAP: Add DSP_A mode support for mcbs
- ASoC: OMAP: Use single-phase for DSP mod
- ASoC: n810: replace BUG() with BUG_ON(
Soc PXA2xx Imote 2
- ASoC: IMote2 ASoC Suppor
Soc PXA2xx Magician
- ASoC: change set_tdm_slot api to allow slot_width override
- ASoC: UDA1380: refactor device registratio
- ASoC: magician: fix PXA SSP clock polarit
- ASoC: Optimize switch/case in magician.
USB
- ALSA: snd_usb_caiaq: add support for Audio2D
USB USX2Y
- Remove multiple KERN_ prefixes from printk format
- ALSA: usx2y - reparent sound devic
USB caiaq
- Clean up useless files and fix .gitignore for caia
- ALSA: snd_usb_caiaq: add support for Audio2D
- ALSA: snd_usb_caiaq: reparent sound devic
- ALSA: snd_usb_caiaq: fix legacy input streamin
- ALSA: snd_usb_caiaq: set mixernam
- ALSA: snd_usb_caiaq: bump version numbe
- ALSA: snd_usb_caiaq: give better shortnam
- ALSA: snd_usb_caiaq: give better longnam
- ALSA: snd_usb_caiaq: use strlcp
- ALSA: snd_usb_caiaq: clean whitespace
USB generic driver
- Fix usbmidi.patc
- regenerate usbaudio.patc
- ALSA: usb-audio - Fix types taken in min(
- sound: usb-audio: do not make URBs longer than sync packet interva
- ALSA: usb-audio - Volume control quirk for QuickCam E 350
- sound: usb-audio: add MIDI drain callbac
- sound: usb-audio: use multiple output URB
- sound: usb-audio: use multiple input URB
- sound: usb-audio: Xonar U1 digital output suppor
- sound: usb-audio: add workaround for Blue Microphones device
- ALSA: usb-audio - Correct bogus volume dB informatio
- ALSA: usb-audio - Use the new TLV_DB_MINMAX typ
- ALSA: usb-audio - rework quirk for TerraTec Aureon USB 5.1 MkI
- trivial: remove extra spac
- ALSA: usb - Add boot quirk for C-Media 6206 USB Audi
- ALSA: usb-audio - errata corrige for quir
- ALSA: usb-audio - Add quirk for Roland/Edirol M-16D
- ALSA: usb-audio - quirk for USB Aureon card
- ALSA: usbaudio - Add delay accoun
- sound: usb-audio: make the MotU Fastlane work agai
Utils
- alsa-info: Version bump to 0.4.5
- alsa-info: use mktemp -
- alsa-info: Check errors from mktem
- alsa-info: revert the behavior of update optio
- alsa-info: Add --output optio
- alsa-info: Fix usage outpu
- alsa-info: Run the new update script automaticall
- alsa-info: Use sysfs if available instead of dmidecod
- alsa-info.sh: include 1 line of dmesg contex
- alsa-info.sh: add dmesg info on ALSA/HD
- alsa-info: Version bump to 0.4.5
- alsa-info.sh: introduce withall(
- alsa-info.sh: let mv fail loudl
- alsa-info.sh: fix whitespace leaked to stdou
- alsa-info.sh: Do not automatically upload alsa inf
- alsa-info.sh: Provide system manufacturer and product name from DM
- Add parsing of def_tristate to mod-dep
VIA82xx driver
- ALSA: via82xx: add option to disable 500ms delay in snd_via82xx_codec_wai
Virtual Master
- ALSA: Add new TLV types for dBwith min/ma
YMFPCI driver
- sound: ymfpci: increase timer resolution to 96 kH
au88x0 driver
- sound: Use PCI_VDEVIC
- ALSA: au88x0: fix wrong period_elapsed() cal
- ALSA: au88x0: fix .pointer callbac
alsa-lib
Core
- Release v1.0.2
- add midi event test
Config API
- fix doc error
- conf.c: more documentatio
Control API
- control.c: snd_ctl_wait: fix revents handlin
- fix doc error
- Add the support of TLV_DB_MINMAX type
- Fix breakage of snd_card_load(
- snd_card_get_index() - extend comment for last chang
- Extend snd_card_get_index() to accept also control device name like /dev/snd/controlC
Mixer API
- remove unimplemented functions from header
PCM API
- pcm/ioplug: fix error code in start callbac
- pcm: workaround for avoiding automatic start in mmap mod
- snd_pcm_scope_set_ops: make ops parameter cons
- Fix zero-division in pcm_rate.
- remove unimplemented functions from header
- pcm_hooks: cosmetic removal of unused variable
- Manage dlobj lifetime in pcm_hooks.
- pcm dmix plugin: fix MIX_AREAS_24 routine for i386 & x86_64 platform
- Query the supported rate ranges from rate plugin
RawMidi API
- sound: rawmidi: disable active-sensing-on-close by defaul
Sequencer API
- more midi_event documentatio
- seq_midi_event: fix decoding of (N)RPN event
- MIDI event decoder: prevent running status after syse
Timer API
- timer_query: make ops structure constan
Configuration
- Fix driver conf parsing in snd_config_hook_load_for_all_cards(
- conf.c: more documentatio
- conf.c: rename 'node' to 'config
- conf.c: rename 'leaf' to 'child
- conf.c: rename 'father' to 'parent
- conf.c: snd_config_add: prevent adopting a non-orpha
- USB-Audio.conf: fix definition for M-Audio AudioPhile spdif devic
- conf.c: fix handling of NULL string value
- conf.c: snd_config_set_id: prevent duplicate id
- conf.c: fix handling of NULL id
- Fix SB-Xfi.con
- Add IEC958 status bits support to SB-XFi.con
- Add config file for SB-XFi drive
Documentation
- doc: hide structs with typedef
- doc: fix handling of @top_srcdir
External PCM I/O Plugin SDK
- fix doc error
External Rate Converter Plugin SDK
- Query the supported rate ranges from rate plugin
I/O subsystem
- fix doc error
Test/Example code
- add config test
- test/lsb/midi_event.c: check for buffer size chec
- test/lsb/midi_event.c: abort on fatal error
- test/pcm.c: float format suppor
- add midi event test
- test/pcm.c: Generic linear PCM suppor
- test/pcm.c: Fix S24 forma
- test/pcm.c: Sample generation on big endian platforms was broken
alsa-utils
Core
- Release v1.0.2
- alsamixer: show channel names for multichannel control
/include/Makefile.am
- alsamixer: show channel names for multichannel control
ALSA Control (alsactl)
- alsactl init rules: fix Lenovo T61 initialization (Speaker Playback Switch
- alsactl: init - fix default configuration for ENS137
- alsactl: fixed Headphone Playback Volume setting in default rule
Speaker Test
- speaker-test: only check byte order onc
- speaker-test: move existing endian macros up in the fil
- Remove dead/commented out cod
- Allow frequencies down to 30 H
- speaker-test: allow frequency to be floating poin
alsamixer
- alsamixer: fix display of inactive volume ba
- alsamixer: rename attr to c
- alsamixer - Tricolorize volume bar
- alsamixer: update man pag
- alsamixer: fix text box clipping with multi-column character
- alsamixer - Fix uninitialized variable warnin
- alsamixer: show channel names for multichannel control
aplaymidi/arecordmidi
- aplaymidi: reduce bandwidth for big SysEx message
alsa-tools
Core
- Release v1.0.2
Envy24 Control
- envy24control - Don't redeclare isblank()
ac3dec (Dolby Digital Decoder)
- ac3dec - Fix typos of -q optio
hdspconf
- Also fix the configure for hdspconf for LIBS/LDFLAGS mistakes
qlo10k1
- qlo10k1: Fix usage of $x_libraries in acinclude.m4 - it may be empt
us428control
- us428control - Fix array overflo
alsa-plugins
Core
- Release v1.0.2
- pulse: use PA_CONTEXT_IS_GOOD where applicabl
Documentation
- speex - Add echo-cancelling option to speexdsp plugi
OSS Mixer -> ALSA Control plugin
- oss - Add missing initialization of fragment
Public Parrot Hack rate converter
- Add PCM rates query support for PCM rate plugin
PulseAudio -> ALSA plugin
- pulse: immediately trigger EIO when connection is droppe
- pulse: rework object destruction paths a bi
- pulse: unify stream/context state check
- pulse: get rid of redundant state variabl
- pulse: move a couple of PCM related functions from pulse.c to pcm_pulse.
- pulse: replace manual mainloop by pa_mainloop_iterate(
- pulse: call pa_threaded_mainloop_wait() to handle spurious wakeup
- pulse: unify destruction of snd_pulse_
- pulse: use PA_CONTEXT_IS_GOOD where applicabl
- pulse: get rid of a number of assert()
- alsa-plugins/pulse: Implement 'pause'
Speex PCM plugin
- speex - Add echo-cancelling option to speexdsp plugi
libavcodec's resampler
- Add PCM rates query support for PCM rate plugin
alsa-python
Core
- Release v1.0.2
- [PATCH] alsa-python: Add support for setuptool
pyalsa.alsaseq module
- pyalsa: fix integer overflow in alsaseq.
- alsaseq: fix time stamp
Detailed changelog between 1.0.20 and 1.0.21 releases
alsa-driver
Sound Core
- - Release v1.0.2
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - Add compat header for linux/regulator/consumer.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Clean up / improve INSTALL documen
- Some more on installation with 2.6.x kernels
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Allow relative path to --with-moddir configure optio
- Allow a relative path to --with-moddir configure option. When a
- relative path is given, the path is appended to /lib/modules/$VERSION/
- For example, the recent module-init-tools prefers the director
- /lib/modules/$VERSION/updates as the update module path to the norma
- directories. Thus, passing --with-moddir=updates will store the newl
- built modules to that directory instead of the standard pat
- /lib/modules/$VERSION/kernel/sound
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add linux/math64.h compat heade
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add check of linux/bug.h in configure scrip
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - sound: make OSS device number claiming optional and schedule its remova
- If any OSS support is enabled, regardless of built-in or module
- sound_core claims full OSS major number (that is, the old 0-25
- region) to trap open attempts and request sound modules using custo
- module aliases. This feature is redundant as chrdev already has suc
- mechanism. This preemptive claiming prevents alternative OS
- implementation
- The custom module aliases are scheduled to be removed and the previou
- patch made soundcore emit the standard chrdev aliases too to hel
- transition
- This patch schedule the feature for removal in a year and makes i
- optional so that developers and distros can try new things in th
- meantime without rebuilding the kernel. The pre-claiming can b
- turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel paramete
- soundcore.preclaim_oss
- As this allows sound minors to be individually grabbed by other users
- this patch updates sound_insert_unit() such that if registerin
- individual device region fails, it tries the next available slot
- For details on removal plan, please read the entry added by this patc
- in feature-removal-schedule.txt
- Signed-off-by: Tejun Heo <tj@kernel.org
- Cc: Alan Cox <alan@lxorguk.ukuu.org.uk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
ALSA Core
- - Add missing definition of KERN_DEFAULT used in misc.c for older kernel
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add compat header for linux/regulator/consumer.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Move the previous hack to adriver.
- It's better to be in adriver.h since config.h is generated
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add a hack to avoid Oops related with jack laye
- Added a hack to avoid Oops in patch_sigmatel.c and patch_conexant.
- due to the wrong kconfigs regarding input jack layer
- Since CONFIG_SND_JACK is enabled only on 2.6.27 or later kernels
- CONFIG_SND_JACK isn't set even though CONFIG_SND_HDA_INPUT_JACK=y
- This causes NULL dereference after snd_jack_new()
- The patch simply disables CONFIG_SND_HDA_INPUT_JACK when CONFIG_SND_JAC
- is undefined, too
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Fix build of hda_intel.
- The commit dc4c2e6bde77735071dbef7aca6bd6c0116102b3 in sound tre
- causes the build errors on older kernels due to undefined PCI id an
- the use of pci_dev.revirsion field. Make a patch to fix the build
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Show the stack trace at bad kfree debug message
- This makes a lot easier to find out the culprit
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add krealloc() workaround for older kernels in core/info.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Use memdup_user() wrapper when memory-debug option is enable
- Use memdup_user() wrapper when memory-debug option is enabled
- Otherwise you'll get "bad kfree()" errors due to mismatching kfree
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add missing PCI_VDEVICE definition for older kernel
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add missing const to memdup_user() wrapper in adriver.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add linux/math64.h compat heade
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ctxfi - Add new PCI ids to pci_ids_compat.h.i
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: Fixed a typo of printk(
- Fixed a silly typo of printk() included in the previous patch..
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: pcm - Increase protocol versio
- Increase the PCM protocol version to indicate the drain ioctl behavio
- change
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Add debug module optio
- Add debug module option to snd core
- This controls the debug print level. When CONFIG_SND_DEBUG_VERBOS
- is set, you can suppress the debug messages by giving or changing thi
- parameter to a lower value. debug=0 means no debug messsages
- As default, it's set to the verbose level 2
- Since this option can be changed dynamically via sysfs file, you ca
- suppress the verbose debug messages on the fly, which wasn't possibl
- before
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: core - strip too long file names in snd_print*(
- When modules are built with M= option, they pass long file paths t
- __FILE__. This results in ugly outputs of snd_print*() whe
- CONFIG_SND_VERBOSE_PRINTK is set
- This patch adds a check of the path and strips the leading path dir
- if the file name is an absolute path to improve the readability of logs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Fix SG-buffer DMA with non-coherent architecture
- Using SG-buffers with dma_alloc_coherent() is often very inefficien
- on non-coherent architectures because a tracking record could b
- allocated in addition for each dma_alloc_coherent() call
- Instead, simply disable SG-buffers but just allocate normal continuou
- buffers on non-supported (currently all but x86) architectures
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: info - Use krealloc(
- Use krealloc() to resize the buffer in sound/core/info.c
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Core - clean up snd_card_set_id* calls and remove possible id collisio
- Move locking outside snd_card_set_id_internal() function and rename i
- to snd_card_set_id_no_lock() for better function description
- User defined id is just copied to card structure at allocation time
- The real unique id procedure is called in snd_card_register() t
- ensure real atomicity
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Fix double locking of card list in snd_card_register(
- The introduction of snd_card_set_id() added a lock on the card lis
- to the old choose_default_id() function when using it to implemen
- the new API call. This lock is needed to allow us to walk the lis
- and check to see if our new name is a duplicate. Unfortunately thi
- causes a lockup when called from snd_card_register() (in case
- where no ID is supplied for the card) since the card list is alread
- locked there
- Fix this fairly hideously by factoring out the implementation an
- using a flag to indicate if the lock should be held. A better fi
- would probably be to refactor snd_card_register() to move th
- _set_id() outside the locking region but I can't immediately se
- anything I can convince myself is safe
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Core - add snd_card_set_id() functio
- Introduce snd_card_set_id() function to allow lowlevel drivers to se
- default identification name for card slot. The function checks als
- for identification name collisions and tries to create unique name
- Also, the snd_card_create() function is simplified, because this ne
- function is used. As bonus, proper name collision checks are evaluate
- at the card create time
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: clean up the logic for building sequencer module
- Instead of mangling the CONFIG_* variables in the makefiles over an
- over, set a few helper variables in Kconfig
- Signed-off-by: Michal Marek <mmarek@suse.cz
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: PCM midlevel: improve fifo_size handlin
- Move the fifo_size assignment to hw->ioctl callback to allow lowleve
- drivers overwrite the default behaviour
- fifo_size is in frames not bytes as specified in asound.h and alsa-lib'
- documentation, but most hardware have fixed byte based FIFOs. Introduc
- internal SNDRV_PCM_INFO_FIFO_IN_FRAMES
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Remove deprecated include/sound/driver.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Remove deprecated snd_card_new(
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC PXA2xx Core
- - ASoC: Pass correct platform data from pxa2xx-ac9
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Allow passing platform_data for pxa2xx-ac9
- This patch adds support for passing platform data to ac97 bus device
- from PXA2xx-AC97 driver.
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: change set_tdm_slot api to allow slot_width override
- Extend set_tdm_slot to allow the user to arbitrarily set the frame widt
- and active TX/RX slots
- Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.
- still doesn't handle the slot_width override
- While being there, correct an incorrect use of SlotsPerFrm(7) use i
- bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) )
- (this series is meant for Mark's for-2.6.32 branch
- Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - [ARM] pxa/palm: Add Palm27x aSoC driver to PalmTE
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Eric Miao <eric.miao@marvell.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: remove duplicated code on pxa-ssp.
- * We don't need to write the registers twice, remove the first write
- * DAIFMT_INV switch is duplicated inside DAIFMT_FORMAT switch, move i
- out
- (This patch is for Mark's for-2.6.32 branch, I have not checked if th
- code is duplicated on current 2.6.30
- Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Only disable pxa2xx-i2s clocks if we enabled the
- The clock API can't cope with unbalanced enables and disables an
- we only enable in hw_params() but try to disable in shutdown
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: pxa2xx-i2s: Fix suspend/resum
- pxa2xx_i2s_resume is
- - unconditionnaly setting SACR0_EN
- - unsetting SACR0_ENB in saved SACR0 pxa_i2s.sacr
- fix these
- In pxa2xx_i2s_{resume,suspend}, save/restore registers eve
- when !dai->active
- Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: pxa2xx-i2s: Fix inappropriate release of i2s cloc
- i2s_clk is 'put' for no reason in pxa2xx_i2s_shutdown
- Now we 'get' i2s_clk at probe and 'put' it at driver removal or whe
- probe fails
- Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: pxa2xx-i2s: Handle SACR1_DRPL and SACR1_DREC separatel
- - hw_params enables both RPL and REC functions each time : Enable th
- appropriate function in pxa2xx_i2s_trigger
- - pxa2xx_i2s_shutdown disables i2s anytime one of RPL or REC function i
- off : Turn it off only when both functions are off
- Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: pxa2xx-i2s: Proper hw initializatio
- Make sure we are in a know good state at end of probe
- Reset FIFO logic and registers, and make sure REC and RPL function
- along with FIFO service are disabled (SACR0_RST enables REC and RPL)
- Resetting loses current settings so remove reset from stream startup
- Now reset occurs only at probe
- Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: pxa2xx-i2s: Proper initializatio
- Reset FIFO logic and registers, and make sure REC and RPL functions alon
- with FIFO service are disabled at probe
- Signed-off-by: Karl Beldan <karl.beldan@mobile-devices.fr
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Enforce symmetric rates for PXA2xx I2
- There is a single I2S_SYNC pin on the chip
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: em-x270: make the driver support also eXeda and CM-X300 machine
- Signed-off-by: Mike Rapoport <mike@compulab.co.il
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: IMote2 ASoC Suppor
- This patch adds the ASoC side of the board support for the Crossbo
- IMB400 daughter board
- Thanks to Crossbow for considerable assistance
- Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: change stereo/mono to 32-bit/16-bit for pxa-ss
- The original idea came from pHilipp, and this makes the code look
- more consistent
- Signed-off-by: Eric Miao <eric.miao@marvell.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: simplify the SSP DMA parameters settings by run-time generatio
- The SSP DMA parameters can actually be easily generated at run-time sinc
- they are almost similar except for the FIFO width and direction. Anothe
- benefit is the re-use of information from 'struct ssp_device', like SSD
- physical FIFO address and DRCMR register index for both directions
- Signed-off-by: Eric Miao <eric.miao@marvell.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Reviewed-by: pHilipp Zabel <philipp.zabel@gmail.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: pxa-ssp.c fix clock/frame inver
- SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low
- SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low
- SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High
- SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High
- SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0)
- This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A an
- DSP_B modes
- Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Control Midlevel
- - sound: snd_ctl_remove_user_ctl: prevent removal of kernel control
- Ensure that userspace can remove only user controls. Controls create
- by kernel drivers must not be removed because they might be reference
- in calls to snd_ctl_notify()
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: snd_ctl_remove_unlocked_id: simplify user control countin
- Move the decrementing of the user controls counter fro
- snd_ctl_elem_remove to snd_ctl_remove_unlocked_id; this saves th
- separate locking of the controls semaphore, and therefore remove
- a harmless race
- Since the purpose of the function is to operate on user controls (th
- control being unlocked is just a prerequisite), rename it t
- snd_ctl_remove_user_ctl
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: snd_ctl_remove_unlocked_id: simplify error path
- Use a common exit path to release the mutex and to return a possibl
- error
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: snd_ctl_elem_add: fix value count chec
- Make sure that no user element that has no values can be added
- The check for count>1024 is not needed because the count is checke
- later for the individual control types
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Add new TLV types for dBwith min/ma
- Add new types for TLV dB scale specified with min/max values instea
- of min/step since the resolution can't match always with the on
- a device provides. For example, usb audio devices give 1/256 d
- resolution while ALSA TLV is based on 1/100 dB resolution
- The new min/max types have less problems because the possibl
- rounding error happens only at min/max
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Jack Input Event Midlevel
- - ALSA: use card device as parent for jack input-device
- This moves the jack devices from the PCI device into the ALSA card device, whic
- makes it easier for userspace to find all devices belonging to a specific car
- while granting access to logged-in users
- Jack input devices from sound cards can now simply be matched with udev by doing
- SUBSYSTEM="input", SUBSYSTEMS="sound", ..
- ls -l /sys/devices/pci0000:00/0000:00:1b.0/sound/card
- controlC
- device -> ../../../0000:00:1b.
- i
- input1
- input1
- input
- input
- numbe
- pcmC0D0
- pcmC0D0
- pcmC0D1
- powe
- subsystem -> ../../../../../class/soun
- ueven
- Cc: Lennart Poettering <lennart@0pointer.de
- Signed-off-by: Kay Sievers <kay.sievers@vrfy.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
PCM Midlevel
- - Refresh pcm_native.patch for drain ioctl fixe
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Regenerate pcm_native.patc
- Also simplify the check of VM_RESERVED
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: pcm - Fix drain behavior in non-blocking mod
- The current PCM core has the following problems regarding PCM drainin
- in non-blocking mode
- - the current f_flags isn't checked in snd_pcm_drain(), thus changin
- the mode dynamically via snd_pcm_nonblock() after open doesn't work
- - calling drain in non-blocking mode just return -EAGAIN error, bu
- doesn't provide any way to sync with draining
- This patch fixes these issues
- - check file->f_flags in snd_pcm_drain() properl
- - when O_NONBLOCK is set, PCM core sets the stream(s) to DRAIN stat
- but quits ioctl immediately without waiting the whole drain; th
- caller can sync the drain manually via poll(
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: pcm - Tell user that stream to be rewound is suspende
- Return STRPIPE instead of EBADF when userspace attempts to rewin
- of forward a stream that was suspended in meanwhile, so that i
- can be recovered by snd_pcm_recover()
- This was causing Pulseaudio to unload the ALSA sink module under a rac
- condition when it attempted to rewind the stream right after resume fro
- suspend, before writing to the stream which would cause it to revive th
- stream otherwise. Tested to work with Pulseaudio patched to attempt t
- snd_pcm_recover() upon receiving an error from snd_pcm_rewind()
- Signed-off-by: Lubomir Rintel <lkundrak@v3.sk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: pcm_lib: fix unsorted list constraint handlin
- snd_interval_list() expected a sorted list but did not document this, s
- there are drivers that give it an unsorted list. To fix this, chang
- the algorithm to work with any list
- This fixes the "Slave PCM not usable" error with USB devices that hav
- multiple alternate settings with sample rates in decreasing order, suc
- as the Philips Askey VC010 WebCam
- http://bugzilla.kernel.org/show_bug.cgi?id=1402
- Reported-and-tested-by: Andrzej <adkadk@gmail.com
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Cc: <stable@kernel.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: pcm - Fix hwptr buffer-size overlap bu
- The fix 79452f0a28aa5a40522c487b42a5fc423647ad98 introduced anothe
- bug due to the missing offset for the overlapped hwptr
- When the hwptr goes back to zero, the delta value has to be correcte
- with the buffer size. Otherwise this causes looping sounds
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: pcm - Fix warnings in debug logging
- Add proper cast
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: pcm - Add logging of hwptr updates and interrupt update
- Added the logging functionality to xrun_debug to record the hwpt
- updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt()
- corresponding to 16 and 8, respectively
- For example
- # echo 9 > /proc/asound/card0/pcm0p/xrun_debu
- will record the position and other parameters at each period interrup
- together with the normal XRUN debugging
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: pcm - Fix regressions with VMwar
- VMware tends to report PCM positions and period updates at utterl
- wrong timing. This screws up the recent PCM core code that trie
- to correct the position based on the irq timing
- Now, when a backward irq position is detected, skip the updat
- instead of rebasing. (This is almost the old behavior befor
- 2.6.30.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Fix SG-buffer DMA with non-coherent architecture
- Using SG-buffers with dma_alloc_coherent() is often very inefficien
- on non-coherent architectures because a tracking record could b
- allocated in addition for each dma_alloc_coherent() call
- Instead, simply disable SG-buffers but just allocate normal continuou
- buffers on non-supported (currently all but x86) architectures
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: fix check for return value in snd_pcm_hw_refin
- 'params' is a pointer and looking at the code this probably should be a chec
- for ioctl return value
- Signed-off-by: Mariusz Kozlowski <m.kozlowski@tuxland.pl
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: pcm - A helper function to compose PCM stream name for debug print
- Use a common helper function for the PCM stream name displayed i
- XRUN and buffer-pointer debug prints
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: pcm - Fix update of runtime->hw_ptr_interrup
- The commit 13f040f9e55d41e92e485389123654971e03b819 made anothe
- regression, the missing update of runtime->hw_ptr_interrupt
- Since this field is only checked in snd_pcmupdate__hw_ptr_interrupt()
- not in snd_pcm_update_hw_ptr(), it must be updated before the hw_pt
- change check
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: Clean up 64bit division function
- Replace the house-made div64_32() with the standard div_u64*() functions
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: PCM midlevel: Fix hw_ptr_jiffies update commi
- In commit "(PCM midlevel: Do not update hw_ptr_jiffies when hw_pt
- is not changed" the hw_ptr change check condition i
- snd_pcm_update_hw_ptr() function was reverted
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: PCM midlevel: lower jiffies check margin using runtime->delay valu
- When hardware has large FIFO, it is necessary to lower jiffies margi
- by count of queued samples
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not change
- Some hardware might have bigger FIFOs and DMA pointer value will be update
- in large chunks. Do not update hw_ptr_jiffies and position timestamp whe
- hw_ptr value was not changed
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: PCM midlevel: introduce mask for xrun_debug() macr
- For debugging purposes, it is better to separate actions
- Bit-values
- 1: show bad PCM ring buffer pointe
- 2: show also stack (to debug kernel latency issues
- 4: check pointer against system jiffie
- Example
- 5: show bad PCM ring buffer pointer and do jiffies chec
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: PCM midlevel: improve fifo_size handlin
- Move the fifo_size assignment to hw->ioctl callback to allow lowleve
- drivers overwrite the default behaviour
- fifo_size is in frames not bytes as specified in asound.h and alsa-lib'
- documentation, but most hardware have fixed byte based FIFOs. Introduc
- internal SNDRV_PCM_INFO_FIFO_IN_FRAMES
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Enable PCM hw_ptr_jiffies check only in xrun_debug mod
- The PCM hw_ptr jiffies check results sometimes in problems when
- hardware doesn't give smooth hw_ptr updates. So far, au88x0 and som
- other drivers appear not working due to this strict check
- However, this check is a nice debug tool, and the capability should b
- still kept
- Hence, we disable this check now as default unless the user enables i
- by setting the xrun_debug mode to the specific stream via a proc file
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Fix invalid jiffies check after paus
- The hw_ptr_jiffies has to be reset properly to avoid the invali
- check of jiffies delta in snd_pcm_update_hw_ptr*() functions
- Especailly this patch fixes the bogus jiffies check after the puas
- and resume
- This patch is a modified version of the original patch by Jaroslav
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Add extra delay count in PC
- Added runtime->delay field to adjust the delayed samples for snd_pcm_delay()
- Typically a hardware FIFO length is stored in this field, so that th
- extra delay between hwptr and applptr can be computed
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
RawMidi Midlevel
- - sound: rawmidi: disable active-sensing-on-close by defaul
- Sending an Active Sensing message when closing a port can interfere wit
- the following data if the port is reopened and a note-on is sent befor
- the device's timeout has elapsed. Therefore, it is better to disabl
- this setting by default
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
T5 and LifeDrive
- - ASoC: Switch palm27x-asoc to jack detection ap
- This patch removes the old method of jack detection from palm27x-aso
- driver and adds jack detection api. It also removes some other (now
- useless stuff from the driver and corrects pin configuration for th
- codec
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - [ARM] pxa/palm: Add Palm27x aSoC driver to PalmTE
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Eric Miao <eric.miao@marvell.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
/include/Makefile
- - Fix mrproper make targe
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
/soc/Makefile
- - Fix build of soc-core.c with older kernel
- Now it's using dev_pm_ops, which was added recently
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: add DMA platform driver for MX1x and MX2
- This adds support for DMA platform valid for i.MX1 and i.MX2 platforms
- This is not valid for i.MX3 since it doesn't share the same DM
- interface than i.MX1 and i.MX2
- It has been tested on i.MX27 board
- Signed-off-by: Javier Martin <javier.martin@vista-silicon.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Begin to factor out register cache I/O function
- A lot of CODECs share the same register data formats and therefor
- replicate the code to manage access to and caching of the registe
- map. In order to reduce code duplication centralised versions o
- this code will be introduced with drivers able to configure the us
- of the common code by calling the new snd_soc_codec_set_cache_io(
- API call during startup
- As an initial user the 7 bit address/9 bit data format used by man
- Wolfson devices is supported for write only CODECs and the driver
- with straightforward register cache implementations are converted t
- use it
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add TXx9 AC link controller driver (v3
- This patch adds support for the integrated ACLC of the TXx9 family
- Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add driver for s6000 I2S interfac
- This patch adds a driver for the I2S interface found on Stretch s600
- family processors
- Signed-off-by: Daniel Glöckner <dg@emlix.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
/soc/codecs/Makefile
- - ASoC: Add ak4642/ak4643 codec suppor
- This is very simple driver for ALS
- It supprt headphone output and stereo input onl
- This patch is tested by ms7724s
- Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out shared code from WM899
- The WM8993 analogue control is shared with other devices in the sam
- product line. Since this is a very substantial proportion of th
- driver move the definitions of these controls into a new wm_hubs modul
- which allows them to be shared between the two
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: new ad1836 codec driver based on aso
- There has been an ad1836 driver in sound/blackfin based on traditional alsa
- The new driver is based on asoc. The architecture of ad1836 codec driver i
- very much like ad1938
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8776 CODEC drive
- The WM8776 is a high performance, stereo audio CODEC with five channe
- input selector. The WM8776 is ideal for surround sound processin
- applications for home hi-fi, DVD-RW and other audio visual equipment
- This driver implements support for most WM8776 features - currently th
- ADC automatic level control/limiter functionality is omitted
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8974 CODEC drive
- The WM8974 is a low power, high quality mono CODEC designed for portabl
- applications such as digital still cameras or digital voice recorders
- This driver was originally written by Graeme Gregory and Liam Girdwoo
- and has since been maintained by myself with some updates contributed b
- Brett Saunders and Javier Martin
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add support for Conexant CX20442-11 voice modem code
- This patch adds support for Conexant CX20442-11 voice modem codec, suitabl
- for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Relate
- sound card driver will follow
- This codec is an optional part of the Conexant SmartV three chip modem design
- As such, documentation for its proprietary digital audio interface is no
- available. However, on Amstrad Delta board, thanks to Mark Underwood wh
- created an initial, omap-alsa based sound driver a few years ago[1], the code
- has been discovered to be accessible not only from the modem side, but als
- over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any soun
- card that can access the codec DAI directly. The DAI configuration parameter
- (sample rate and format, number of channels) has been selected out empiricall
- for best user experience
- The codec analogue interface consists of two pairs of analogue I/O pins
- speakerphone interface or telephone handset/headset interface. Furthermore, i
- seams to provide two operation modes for speakerphone I/O: standard an
- advanced, with automatic gain control and echo cancelation. Even if the code
- control interface is unknown and not available, all those interfaces and mode
- can be selected over the modem chip using V.253 commands. The driver is abl
- to issue necessary commands over a suitable hw_write function if provided by
- sound card driver. Otherwise, the codec can be controlled over the modem fro
- userspace while inactive
- Even if nothig is known about the codec internal power managemen
- capabilities, DAPM widgets has been used to model the codec audio map
- Automatically performed powering up/down of those virtual widgets results i
- corresponding V.253 commands being issued
- Some driver features/oddities may be board specific, but I have no way t
- verify that with any board other than Amstrad Delta
- [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.htm
- Created and tested against linux-2.6.31-rc3
- Applies and works with linux-omap-2.6 commi
- 7c5cb7862d32cb344be7831d466535d5255e35ac as well
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: new ad1938 codec driver based on aso
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: MAX9877: add MAX9877 amp drive
- The MAX9877 combines a high-efficiency Class D audio power amplifie
- with a stereo Class AB capacitor-less DirectDrive headphone amplifier
- The max9877_add_controls() is called to register the MAX9877 specifi
- controls on machine specific init() of the machine driver
- The datasheet for the MAX9877 can find at the following url
- http://datasheets.maxim-ic.com/en/ds/MAX9877.pd
- [Slight edit to sort the ALL_CODECS entries -- broonie.
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8993 CODEC drive
- The WM8993 is a highly integrated ultra-low power hi-fi CODEC designe
- for portable devices such as multimedia phones
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8523 CODEC drive
- The WM8523 is a high performance stereo DAC with integral charg
- pump providing 2Vrms line driver outputs using a single 3.3V powe
- supply rail
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8961 drive
- The WM8961 is a low power, high quality stereo CODEC designed fo
- portable digital applications with headphone and stereo class D speake
- drivers
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add dummy S/PDIF codec suppor
- McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed
- This patch provides stub codec that can be used in these configurations
- On DM646x EVM the McASP1 is connected to the S/PDIF out
- Signed-off-by: Steve Chen <schen@mvista.com
- Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com
- Signed-off-by: Naresh Medisetty <naresh@ti.com
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Codec for STAC9766 used on the Efik
- Datasheet: http://www.idt.com/products/getDoc.cfm?docID=1313400
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier drive
- The WM9081 is designed to provide high power output at low distortio
- levels in space-constrained portable applications
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: ASoC WM8940 Drive
- Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8960 CODEC drive
- The WM8960 is a low power, high quality stereo codec designed fo
- portable digital audio applications
- Stereo class D speaker drivers provide 1W per channel into 8W loads
- Guaranteed low leakage, excellent PSRR and pop/click suppressio
- mechanisms enable direct battery connection for the speaker supply
- The device also integrates a complete microphone interface and a stere
- headphone driver. External component requirements are drasticall
- reduced as no separate microphone, speaker or headphone amplifiers ar
- required. Advanced on-chip digital signal processing performs automati
- level control for the microphone or line input
- Stereo 24-bit sigma-delta ADCs and DACs are used with low powe
- over-sampling digital interpolation and decimation filters and
- flexible digital audio interface
- The master clock can be input directly or generated internally by a
- onboard PLL, supporting most commonly-used clocking schemes
- This driver was originally written by Liam Girdwood, with substantia
- subsequent additions and updates for feature completeness and changes i
- the ASoC framework from me
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8988 CODEC drive
- The WM8988 is a low power, high quality stereo CODEC designed fo
- portable digital audio applications
- The device integrates complete interfaces to 2 stereo headphone or lin
- out ports. External component requirements are drastically reduced as n
- separate headphone amplifiers are required. Advanced on-chip digita
- signal processing performs graphic equaliser, 3-D sound enhancement an
- automatic level control for the microphone or line input
- The WM8988 can operate as a master or a slave, with various master cloc
- frequencies including 12 or 24MHz for USB devices, or standard 256f
- rates like 12.288MHz and 24.576MHz. Different audio sample rates such a
- 96kHz, 48kHz, 44.1kHz are generated directly from the master cloc
- without the need for an external PLL
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
/soc/pxa/Makefile
- - ASoC: IMote2 ASoC Suppor
- This patch adds the ASoC side of the board support for the Crossbo
- IMB400 daughter board
- Thanks to Crossbow for considerable assistance
- Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
AC97 Codec
- - ALSA: Allow passing platform_data for pxa2xx-ac9
- This patch adds support for passing platform data to ac97 bus device
- from PXA2xx-AC97 driver.
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Allow passing platform_data to devices attached to AC97 bu
- This patch allows passing platform_data to devices attached to AC97 bu
- (like touchscreens, battery measurement chips ...)
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Kill truncate warning by shortening Sigmatel-specific AC97 control nam
- ALSA sound/core/control.c:232: Control name 'Sigmatel Surround Phas
- Inversion Playback Switch' truncated to 'Sigmatel Surround Phas
- Inversion Playback ' bootup message by omitting weird Sigmatel prefi
- in this case; also fix up the related ca0106 mixer control remova
- part by using identical naming there
- Signed-off-by: Andreas Mohr <andi@lisas.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
ALI5451 driver
- - ALSA: ali5451: remove dead cod
- Remove code covered by #if/endif 0 and #ifdef/endif CODEC_RESE
- (CODEC_RESET is never defined)
- Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready(
- Modify loops in such way that the register value is checked also afte
- the timeout condition, just in case the heavy interrupt load etc. cause
- the thread to sleep for the time period exceeding the timeout value
- While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready()
- Reported-by: Jack Byer <ojbyer@usa.net
- Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
ALSA sequencer
- - ALSA: OSS sequencer should be initialized after snd_seq_system_client_ini
- When build SND_SEQUENCER in kernel then OSS sequencer(alsa_seq_oss_init
- is initialized before System (snd_seq_system_client_init) which leads t
- memory leak
- unreferenced object 0xf6b0e680 (size 256)
- comm "swapper", pid 1, jiffies 429467075
- backtrace
- [<c108ac5c>] create_object+0x135/0x20
- [<c108adfe>] kmemleak_alloc+0x26/0x4
- [<c1087de2>] kmem_cache_alloc+0x72/0xf
- [<c126d2ac>] seq_create_client1+0x22/0x16
- [<c126e3b6>] snd_seq_create_kernel_client+0x72/0xe
- [<c1485a05>] snd_seq_oss_create_client+0x86/0x14
- [<c1485920>] alsa_seq_oss_init+0xf6/0x15
- [<c1001059>] do_one_initcall+0x4f/0x11
- [<c14655be>] kernel_init+0x115/0x16
- [<c10032af>] kernel_thread_helper+0x7/0x1
- [<ffffffff>] 0xfffffff
- unreferenced object 0xf688a580 (size 64)
- comm "swapper", pid 1, jiffies 429467075
- backtrace
- [<c108ac5c>] create_object+0x135/0x20
- [<c108adfe>] kmemleak_alloc+0x26/0x4
- [<c1087de2>] kmem_cache_alloc+0x72/0xf
- [<c126f964>] snd_seq_pool_new+0x1c/0xb
- [<c126d311>] seq_create_client1+0x87/0x16
- [<c126e3b6>] snd_seq_create_kernel_client+0x72/0xe
- [<c1485a05>] snd_seq_oss_create_client+0x86/0x14
- [<c1485920>] alsa_seq_oss_init+0xf6/0x15
- [<c1001059>] do_one_initcall+0x4f/0x11
- [<c14655be>] kernel_init+0x115/0x16
- [<c10032af>] kernel_thread_helper+0x7/0x1
- [<ffffffff>] 0xfffffff
- unreferenced object 0xf6b0e480 (size 256)
- comm "swapper", pid 1, jiffies 429467075
- backtrace
- [<c108ac5c>] create_object+0x135/0x20
- [<c108adfe>] kmemleak_alloc+0x26/0x4
- [<c1087de2>] kmem_cache_alloc+0x72/0xf
- [<c12725a0>] snd_seq_create_port+0x51/0x21
- [<c126de50>] snd_seq_ioctl_create_port+0x57/0x13
- [<c126d07a>] snd_seq_do_ioctl+0x4a/0x6
- [<c126d0de>] snd_seq_kernel_client_ctl+0x33/0x4
- [<c1485a74>] snd_seq_oss_create_client+0xf5/0x14
- [<c1485920>] alsa_seq_oss_init+0xf6/0x15
- [<c1001059>] do_one_initcall+0x4f/0x11
- [<c14655be>] kernel_init+0x115/0x16
- [<c10032af>] kernel_thread_helper+0x7/0x1
- [<ffffffff>] 0xfffffff
- The correct order should be
- System (snd_seq_system_client_init) should be initialized befor
- OSS sequencer(alsa_seq_oss_init) which is equivalent to
- 1. insmod sound/core/seq/snd-seq-device.k
- 2. insmod sound/core/seq/snd-seq.k
- 3. insmod sound/core/seq/snd-seq-midi-event.k
- 4. insmod sound/core/seq/oss/snd-seq-oss.k
- Including sound/core/seq/oss/Makefile after other seq module
- fixes the ordering and memory leak
- Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - sound: rawmidi: disable active-sensing-on-close by defaul
- Sending an Active Sensing message when closing a port can interfere wit
- the following data if the port is reopened and a note-on is sent befor
- the device's timeout has elapsed. Therefore, it is better to disabl
- this setting by default
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: seq_midi: do not send MIDI reset when closin
- Sending a MIDI reset message when closing a port is wrong because w
- only want to shut the device up, not to reset all settings
- Furthermore, many devices ignore this message
- Fortunately, the RawMIDI layer already shuts the device up, so we ca
- ignore this matter here
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: seq-midi: always log message on output overru
- It turns out that the main cause of output buffer overruns is not slo
- drivers but applications that generate too many messages. Therefore, i
- makes more sense to make that error message always visible, and t
- rate-limit it
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: seq_midi_event: fix decoding of (N)RPN event
- When decoding (N)RPN sequencer events into raw MIDI commands, th
- extra_decode_xrpn() function had accidentally swapped the MSB and LS
- controller values of both the parameter number and the data value
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Cc: <stable@kernel.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: clean up the logic for building sequencer module
- Instead of mangling the CONFIG_* variables in the makefiles over an
- over, set a few helper variables in Kconfig
- Signed-off-by: Michal Marek <mmarek@suse.cz
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
ALSA<-OSS emulation
- - ALSA: Clean up 64bit division function
- Replace the house-made div64_32() with the standard div_u64*() functions
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
ALSA<-OSS sequencer
- - sound: seq_oss_midi: remove magic number
- Instead of using magic numbers for the controlles sent when resettin
- a port, use the symbols from asoundef.h
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
ARM AACI PL041 driver
- - [ARM] 5544/1: Trust PrimeCell resource size
- I found the PrimeCell/AMBA Bus drivers distrusting the resourc
- passed in as part of the struct amba_device abstraction. Thi
- patch removes all hard coded resource sizes found in the PrimeCel
- drivers and move the responsibility of this definition back t
- the platform/board device definition, which already exist an
- appear to be correct for all in-tree users of these drivers
- We do this using the resource_size() inline function which wa
- also replicated in the only driver using the resource size, s
- that has been changed too. The KMI_SIZE was left in kmi.h in cas
- someone likes it. Test-compiled against Versatile and Integrato
- defconfigs, seems to work but I don't posess these boards an
- cannot test them
- Signed-off-by: Linus Walleij <linus.walleij@stericsson.com
- Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk
- - [ARM] 5519/1: amba probe: pass "struct amba_id *" instead of void
- The second argument of the probe method points to the amba_i
- structure, so it's better passed with the correct type. None of th
- current in-tree drivers uses the pointer, so they have only bee
- checked for a clean compile
- Change suggested by Russell King
- Signed-off-by: Alessandro Rubini <rubini@unipv.it
- Signed-off-by: Russell King <rmk+kernel@arm.linux.org.uk
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
ARM PXA2XX driver
- - ASoC: Pass correct platform data from pxa2xx-ac9
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Restore support for DMAless DAIs on PX
- Used for applications such as direct bluetooth connections o
- smartphones which don't go via the CPU. This used to be supporte
- before the refactoring to share code but this check was remove
- during that move
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Allow passing platform_data for pxa2xx-ac9
- This patch adds support for passing platform data to ac97 bus device
- from PXA2xx-AC97 driver.
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix NULL pointer dereference in __pxa2xx_pcm_hw_fre
- Check for rtd->params->drcmr != NULL before accessing it
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - pxa2xx-ac97: fix reset gpio mode settin
- Signed-off-by: Mike Rapoport <mike@compulab.co.il
- Acked-by: Eric Miao <eric.miao@marvell.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
ATIIXP driver
- - sound: Use PCI_VDEVIC
- Signed-off-by: Joe Perches <joe@perches.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
ATIIXP-modem driver
- - sound: Use PCI_VDEVIC
- Signed-off-by: Joe Perches <joe@perches.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
AZT3328 driver
- - ALSA: azt3328: fix previous breakage, improve suspend, cleanup
- - fix my previous codec activity breakage (_non-warned_ variable assignmen
- issue
- - convert suspend/resume to 32bit I/O access (I/O is painful; to improv
- suspend/resume performance
- - change DEBUG_PLAY_REC to DEBUG_CODEC for consistenc
- - printk cleanu
- - some logging improvement
- - minor cleanup/improvement
- The variable assignment issue above was a conditional assignment to th
- call_function variable (this ended with the non-preinitialized variabl
- not getting assigned in some cases, thus a dangling stack value, yet gcc 4.3.
- unbelievably did _NOT_ warn about it in this case!!)
- needed to change this into _always_ assigning the check result
- Practical result of this bug was that when shutting dow
- _either_ playback or capture, _both_ streams dropped dead :
- Tested, working (plus resume) and checkpatch.pl:ed on 2.6.30-rc5
- applies cleanly to 2.6.30 proper with my previous (committed
- patches applied
- Signed-off-by: Andreas Mohr <andi@lisas.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: azt3328: large codec cleanup, add I2S port etc
- - fully separate codec I/O port handling, enabling the use of a singl
- function each for all codecs (playback, capture, I2S out
- - add a new separate pcm for I2S out port (UNTESTED, no I2S DA
- available yet
- - switch gameport to low frequency while idle, to try to reduce noise/powe
- - improve snd_azf3328_codec_setdmaa() calculatio
- - minor variable type cleanup (u16, bool etc.
- - add some doc updates (help those lost Windows users, debug help, ...
- Note that due to the large cleanup aspect of the codec I/O change
- I was able to fit everything including all improvements into th
- same binary size!! (a measly 10 bytes more or so
- This should now be the almost last patch to this drive
- (minus some possible kernel clocksource patch and x86_64 fixes or so)
- I just felt like taking a break from the usual stuff and wanted t
- get this driver's structure finished, and it's rather clean now..
- Tested, working and checkpatch.pl:ed on 2.6.30-rc5
- applies cleanly to 2.6.30 proper
- Signed-off-by: Andreas Mohr <andi@lisas.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Apple Onboard Audio driver
- - ALSA: sound/aoa: Add kmalloc NULL test
- Check that the result of kzalloc is not NULL before a dereference
- The semantic match that finds this problem is as follows
- (http://www.emn.fr/x-info/coccinelle/
- // <smpl
- @
- expression *x
- identifier f
- constant char *C
- @
- x = \(kmalloc\|kcalloc\|kzalloc\)(...)
- ... when != x == NUL
- when != x != NUL
- when != (x || ...
- kfree(x
- f(...,C,...,x,...
- *f(...,x,...
- *x->
- // </smpl
- Signed-off-by: Julia Lawall <julia@diku.dk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - sound: remove driver_data direct access of struct devic
- In the near future, the driver core is going to not allow direct acces
- to the driver_data pointer in struct device. Instead, the function
- dev_get_drvdata() and dev_set_drvdata() should be used. These function
- have been around since the beginning, so are backwards compatible wit
- all older kernel versions
- Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de
- Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Au12x0/Au1550 PSC ASoC
- - Add missing ASoC build stub
- Signed-off-by: Takashi Iwai <tiwai@suse.de
BT87x driver
- - ALSA: bt87x - Add a quirk entry for Askey Computer Corp. MagicTView'9
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some driver
- Added SNDRV_PCM_INFO_BATCH flag to PCM info field of some drivers tha
- really don't give the precise pointer value
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
CA0106 driver
- - ALSA: ca0106 - Fix the max capture buffer siz
- The capture buffer size with 64kB seems broken with CA0106
- At least, either the update timing or the DMA position is wrong
- and this screws up pulseaudio badly
- This patch restricts the max buffer size less than that to make lif
- a bit easier
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Cc: <stable@kernel.org
- - sound: Use PCI_VDEVICE for CREATIVE and ECTIV
- Here's a patch on top of the others to use CREATIVE and ECTIV
- Signed-off-by: Joe Perches <joe@perches.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ca0106 - Fix master volume scal
- The master volume dB scale was wrongly defined as 0.50dB setp whil
- it must be 0.25dB step
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ca0106 - Add missing card->mixername field setu
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Remove invalid GENERIC_MIX PCM sublas
- SNDRV_PCM_SUBCLASS_GENERIC_MIX is mostly for h/w multi-stream playbac
- devices, but ca0106 and emu10k1x don't support it (unlike emu10k1)
- We shouldn't set that flag to avoid confusion
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ca0106 - Add missing registrations of vmaster control
- Although the vmaster controls are created, they aren't registered thu
- they don't appear in the real world. Added the missing snd_ctl_add(
- calls
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Cc: <stable@kernel.org
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Kill truncate warning by shortening Sigmatel-specific AC97 control nam
- ALSA sound/core/control.c:232: Control name 'Sigmatel Surround Phas
- Inversion Playback Switch' truncated to 'Sigmatel Surround Phas
- Inversion Playback ' bootup message by omitting weird Sigmatel prefi
- in this case; also fix up the related ca0106 mixer control remova
- part by using identical naming there
- Signed-off-by: Andreas Mohr <andi@lisas.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
CMI8330 driver
- - ALSA: cmi8330: Allow MPU-401-less operatio
- Adding MPU-401 support to cmi8330 driver could cause a regression (non-workin
- sound) on a system where there is no free IRQ for the MPU-401 device (whic
- is not very uncommon as this card requires two separate IRQs plus a third on
- for MPU-401)
- When MPU-401 PnP configuration fails (mostly because of unavailable IRQ), jus
- ignore MPU-401 and continue without it
- Signed-off-by: Ondrej Zary <linux@rainbow-software.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: cmi8330: find OPL3 port automaticall
- My CMI8329 had OPL3 port specified in SB16 resources. But now I found out tha
- it was my modification of the card's PnP EEPROM a couple of years ago (can b
- done using C9SETROM.EXE utility). I did it because the OPL3 port wa
- completely missing from PnP data. It seems to be hardwired to 0x388 o
- CMI8329
- Find OPL3 port automatically by searching in WSS and SB16 resources. If no
- found, assume that it's hardwired to 0x388
- Signed-off-by: Ondrej Zary <linux@rainbow-software.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: cmi8330: Add basic CMI8329 suppor
- Add basic support for CMI8329 cards. Makes PCM and OPL3 work
- Does not break CMI8330 (tested)
- Signed-off-by: Ondrej Zary <linux@rainbow-software.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: cmi8330: revert comments about AD1848 bac
- In ALSA 1.0.20, the comments were changed to say CMI8330 instead of AD1848
- The CMI8330 chip includes two codecs - AD1848 and SB16, so the comments wer
- correct and are misleading now. Revert them back
- Signed-off-by: Ondrej Zary <linux@rainbow-software.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: cmi8330: fix MPU-401 PnP init copy&paste bu
- Fix copy&paste bug in PnP MPU-401 initialization
- Signed-off-by: Ondrej Zary <linux@rainbow-software.org
- Cc: <stable@kernel.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
CMI8788 (Oxygen) driver
- - sound: virtuoso: fix Xonar D1/DX silence after resum
- When resuming, we better take the DACs out of the reset state befor
- trying to use them
- Reference: kernel bug #1359
- http://bugzilla.kernel.org/show_bug.cgi?id=1359
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Cc: <stable@kernel.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - sound: oxygen: make mic volume control mon
- The microphone input and its volume register have only one channel, s
- we have to make the corresponding mixer control a mono control
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - sound: virtuoso: add Xonar Essence ST suppor
- Add support for the Asus Xonar Essence ST and its daughterboard
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - sound: virtuoso: enable HDAV S/PDIF inpu
- The Xonar HDAV1.3 has a digital input jack, so enable the correspondin
- device
- This is not related to the HDMI stuff, which stays unsupported
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - sound: virtuoso: add another DX PCI I
- Add another PCI ID for a second revision of the Xonar DX
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - sound: oxygen: reset DMA when stream is close
- When a PCM stream is closed, flush the corresponding DMA channel
- Otherwise, the DMA controller would continue to output the last sampl
- which would result in a DC offset on the output
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
CMIPCI driver
- - sound: Use PCI_VDEVIC
- Signed-off-by: Joe Perches <joe@perches.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
Conexant Riptide driver
- - Regenerated riptide.patc
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: riptide - proper handling of pci_register_driver for joystic
- We need to check returning error for pci_register_driver(&joystick_driver
- On failure, we should unregister formerly registered audio driver
- This also fixed the compiler warning
- CC [M] sound/pci/riptide/riptide.
- sound/pci/riptide/riptide.c: In function ‘alsa_card_riptide_init’
- sound/pci/riptide/riptide.c:2200: warning: ignoring return value of ‘__pci_register_driver’, declared with attribute warn_unused_resul
- Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: riptide - Fix joystick resource handlin
- The current code doesn't handle the multiple gameports properly
- and uses unnecessary global static variables to store the data
- This patch changes the probe / remove routines to use the drive
- data assigned to the dedicated pci device, and adds the support o
- multiple devices
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: riptide - Code clean u
- A code clean up, coding style fixes
- The firmware loading routine is split to an own function to improv
- the readability
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: riptide: postfix increment and off by on
- With a postfix increment these variables are incremented beyon
- CMDIF_TIMEOUT / MAX_WRITE_RETRY
- Signed-off-by: Roel Kluin <roel.kluin@gmail.com
- Signed-off-by: Andrew Morton <akpm@linux-foundation.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
Creative Sound Blaster X-Fi (20K1/20K2)
- - Fix ctatc.patc
- Regenrated
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add missing pci/ctxfi/cttimer.
- - ctxfi - Fix build with older kerne
- Fix pci->revision for older kernel (to use snd_pci_revision() macro
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add snd-ctxfi build stu
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Simple code clean u
- - replace NULL == xxx with !xx
- - replace NULL != xxx with xx
- - similar trivial cleanup
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Fix uninitialized error check
- Fix a few uninitialized error checks that were introduced recentl
- mistakenlly during the clean-up
- sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’
- sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this functio
- sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’
- sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this functio
- sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’
- sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this functio
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Native timer support for emu20k
- Added the native timer support for emu20k2, which gives much mor
- accurate update timing than the system timer
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi: Swapped SURROUND-SIDE channels on emu20k
- On Soundblaster X-FI Titanium with emu20k2 the SIDE and SURROUN
- channels were swapped and wrong
- I double checked it with connector colors and creative soundblaste
- windows drivers
- So I swapped them to the true order
- Now "speaker-test -c6" and "speaker-test -c8" are working fine
- Signed-off-by: Frank Roth <frashman@freenet.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Add PM suppor
- Added the suspend/resume support to ctxfi driver
- The team tested on the following seems ok
- AMD Athlon 64 3500+ / ASUS A8N-E / 512MB DDR ATI / Radeon X130
- 20k1 & 20k2 card
- Signed-off-by: Wai Yew CHAY <wychay@ctl.creative.com
- Singed-off-by: Ryan RICHARDS <ryan_richards@creativelabs.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Allow unknown PCI SSID
- Allow unknown PCI SSIDs for emu20k1 and emu20k2 as "unknown" model
- Also, add a black-list check in case any device has to be liste
- as "unsupported". It has a negative value in the pci quirk entry
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Fix deadlock with xfi-time
- The PCM x-fi native update routine can cause deadlocks when th
- trigger(START) is called while the stream is running
- This patch fixes the deadlock by just postponing the pcm period updat
- to the next possible wake-up. Also it adds the flip of ti->runnin
- flag (just to be sure as now)
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Replace atc lock to mute
- The spinlock in atc can cause a sleep in lock
- Kernel failure message 1
- BUG: sleeping function called from invalid context at mm/slub.c:159
- in_atomic(): 0, irqs_disabled(): 1, pid: 2537, name: gstreamer-prop
- Pid: 2537, comm: gstreamer-prope Tainted:
- 2.6.29.4-167.fc11.x86_64 #
- Call Trace
- [<ffffffff8103ff0f>] __might_sleep+0x10b/0x11
- [<ffffffff810cd734>] __kmalloc+0x73/0x13
- [<ffffffffa0b4b142>] ? daio_rsc_init+0xaa/0x125 [snd_ctxfi
- [<ffffffffa0b4b212>] dao_rsc_init+0x55/0x1c0 [snd_ctxfi
- [<ffffffffa0b4b3d2>] dao_rsc_reinit+0x55/0x5d [snd_ctxfi
- [<ffffffff813abd6c>] ? _spin_lock_irqsave+0x32/0x3
- [<ffffffffa0b454fe>] atc_spdif_out_passthru+0x92/0x136 [snd_ctxfi
- ..
- Since the lock path is no critical path, it can be gracefull
- replaced with a mutex
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Clear PCM resources at hw_params and hw_fre
- Currently the PCM resources are allocated only once and ever in prepar
- callback, assuming that the PCM parameters are never changed. But it'
- not true
- This patch adds the call of atc->pcm_release_resources() at hw_param
- and hw_free callbacks to assure that the PCM setup is done correctl
- for each h/w parameter changes
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Check the presence of SRC instance in PCM pointer callback
- The SRC instances may not exist when PCM pointer callback is called a
- the state before initialization is finished. Add the NULL check jus
- to be sure
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Add missing start check in atc_pcm_playback_start(
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Add use_system_timer module optio
- Added use_system_timer module option to force to use the system time
- instead of emu20k1 timer irq for debugging
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Fix wrong model id for UA
- CTUAA should be checked instead of CTHENDRIX. The latter is for 20k2 chip
- Also, fixed the detection of UAA/HENDRIX models by fixing the mask bits
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Clean up probe routine
- Clean up probe routines and model detection routines so that the drive
- won't call and check the PCI subsystem id at each time
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Fix / clean up hw20k2 chip cod
- - Clean up Hungarian coding styl
- - Don't use static variables for I2C information; this unables to us
- multiple instances. Now they are stored in struct hw20k2 fields
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Fix possible buffer pointer overru
- Fix possible buffer pointer overruns. Back to zero when it's equa
- or over the buffer size
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Remove useless initializations and cas
- Remove useless variable initializations and cast at the beginning o
- functions
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Fix DMA mask for emu20k2 chi
- Allow 64bit DMA mask for emu20k2 chip, too
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Make volume controls more intuitiv
- Change the volume control to dB scale (as the raw data seems so)
- Also added the TLV dB-scale information
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Optimize the native timer handling using wc counte
- Optimize the timer update routine to look up wall clock once instead o
- checking the position of each stream at each timer update
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ctxfi - Add missing inclusion of linux/math64.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Set device 0 for mixer control element
- Mixer control elements are usually assigned to device 0
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Clean up / optimiz
- - Use static tables instead of assigining each funciton pointe
- - Add __devinit* to appropriate places; pcm, mixer and timer cannot b
- marked because they are kept in the function table that lives lon
- - Move create_alsa_devs function out of struct ct_atc to mark i
- __devini
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Set periods_min to
- Set 2 to minimal periods of playback pcm setups, too
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Use native timer interrupt on emu20k
- emu20k1 has a native timer interrupt based on the audio clock, whic
- is more accurate than the system timer (from the synchronization POV)
- This patch adds the code to handle this with multiple streams
- The system timer is still used on emu20k2, and can be used also fo
- emu20k1 easily by changing USE_SYSTEM_TIMER to 1 in cttimer.c
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Fix previous fix for 64bit DM
- Remove unneeded substitution to 32bit int to make it really working
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Fix endian-dependent code
- The UAA-mode check in hwct20k1.c is implemented with the endian-dependen
- codes. Fix to be more portable (and readable)
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Allow 64bit DM
- emu20kx chips support 64bit address PTE. Allow the DMA bit mask t
- accept 64bit address, too
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Support SG-buffer
- Use SG-buffers instead of contiguous pages
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Remove PAGE_SIZE limitatio
- Remove the limitation of PAGE_SIZE to be 4k by defining the ow
- page size and macros for 4k. 8kb page size could be natively supported
- but it's disabled right now for simplicity
- Also, clean up using upper_32_bits() macro
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Fix supported PCM format
- The device seems supporting only U8, S16, S24_3LE, S32. Other linea
- formats result in bad outputs
- Also, added the support for 32bit float format, which wasn't liste
- in the original code
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Fix PCM device namin
- PCM names for surround streams should be also fixed as well as the mixe
- element names. Also, a bit clean up for PCM name setup
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Fix surround mixer name
- We usually pick up "Surround" mixer for the rear output, and "Side
- for the extra surround. Fix the channel mapping to follow it
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ALSA: ctxfi - Release PCM resources at each prepare cal
- The prepare callback can be called multiple times, thus it needs t
- release and acquire the resource again by itself at the second or late
- call
- Simply add pcm_release_resources() at the beginning of each prepar
- callback in ctatc.c
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Fix Oops at mmappin
- Replace a spinlock with a mutex protecting the vm block list a
- mmap / munmap calls, which caused Oops like below
- BUG: sleeping function called from invalid context at mm/slub.c:159
- in_atomic(): 0, irqs_disabled(): 1, pid: 32065, name: xin
- Pid: 32065, comm: xine Tainted: P 2.6.29.4-75.fc10.x86_64 #
- Call Trace
- [<ffffffff81040685>] __might_sleep+0x105/0x10
- [<ffffffff810c9fae>] kmem_cache_alloc+0x32/0xe
- [<ffffffffa08e3110>] ct_vm_map+0xfa/0x19e [snd_ctxfi
- [<ffffffffa08e1a07>] ct_map_audio_buffer+0x4c/0x76 [snd_ctxfi
- [<ffffffffa08e2aa5>] atc_pcm_playback_prepare+0x1d7/0x2a8 [snd_ctxfi
- [<ffffffff8105ef3f>] ? up_read+0x9/0x
- [<ffffffff81186b61>] ? __up_read+0x7c/0x8
- [<ffffffffa08e36a6>] ct_pcm_playback_prepare+0x39/0x60 [snd_ctxfi
- [<ffffffffa0886bcb>] snd_pcm_do_prepare+0x16/0x28 [snd_pcm
- [<ffffffffa08867c7>] snd_pcm_action_single+0x2d/0x5b [snd_pcm
- [<ffffffffa08881f3>] snd_pcm_action_nonatomic+0x52/0x6a [snd_pcm
- [<ffffffffa088a723>] snd_pcm_common_ioctl1+0x404/0xc79 [snd_pcm
- [<ffffffff810c52c8>] ? alloc_pages_current+0xb9/0xc
- [<ffffffff810c9402>] ? new_slab+0x1a5/0x1c
- [<ffffffff810ab9ea>] ? vma_prio_tree_insert+0x23/0xc
- [<ffffffffa088b411>] snd_pcm_playback_ioctl1+0x213/0x230 [snd_pcm
- [<ffffffff810b6c20>] ? mmap_region+0x397/0x4c
- [<ffffffffa088bd9b>] snd_pcm_playback_ioctl+0x2e/0x36 [snd_pcm
- [<ffffffff810ddc64>] vfs_ioctl+0x2a/0x7
- [<ffffffff810de130>] do_vfs_ioctl+0x462/0x4a
- [<ffffffff81029cef>] ? default_spin_lock_flags+0x9/0x
- [<ffffffff81374647>] ? trace_hardirqs_off_thunk+0x3a/0x6
- [<ffffffff810de1c5>] sys_ioctl+0x55/0x7
- [<ffffffff8101133a>] system_call_fastpath+0x16/0x1
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Fix a typo in MODULE_LICENS
- A space has to be put between GPL and v2
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Add missing module parameter definition
- Added missing module_param*() and MODULE_PARM*()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Move PCI ID definitions to linux/pci_ids.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Add missing inclusion of linux/delay.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Avoid unneeded pci_read_config_*() call
- Use struct pci subsystem_device and revision fields instead o
- unneeded calls of pci_read_config_*()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Add prefix to debug print
- Added ctxfi: prefix to each debug print
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: SB X-Fi driver merg
- The Sound Blaster X-Fi driver supports Creative solutions based o
- 20K1 and 20K2 chipsets
- Supported hardware
- Creative Sound Blaster X-Fi Titanium Fatal1ty® Champion Serie
- Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Serie
- Creative Sound Blaster X-Fi Titanium Professional Audi
- Creative Sound Blaster X-Fi Titaniu
- Creative Sound Blaster X-Fi Elite Pr
- Creative Sound Blaster X-Fi Platinu
- Creative Sound Blaster X-Fi Fatal1t
- Creative Sound Blaster X-Fi XtremeGame
- Creative Sound Blaster X-Fi XtremeMusi
- Current release features
- * ALSA PCM Playbac
- * ALSA Recor
- * ALSA Mixe
- Note
- * External I/O modules detection not included
- Signed-off-by: Wai Yew CHAY <wychay@ctl.creative.com
- Singed-off-by: Ryan RICHARDS <ryan_richards@creativelabs.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Digigram VX222 driver
- - sound: vx222: fix input level control range chec
- Fix a logic error in the range check of the input level control tha
- would prevent setting any volume less than the maximum
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - trivial: fix typo milisecond/millisecond for documentation and source comments
- Signed-off-by: Martin Olsson <martin@minimum.se
- Signed-off-by: Jiri Kosina <jkosina@suse.cz
Documentation
- - ALSA: hda - Add / fix model entries for HD-audio drive
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add quirk for MacBook Pro 5,5 with CS420
- Add the default pin configs for MBP55
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Add debug module optio
- Add debug module option to snd core
- This controls the debug print level. When CONFIG_SND_DEBUG_VERBOS
- is set, you can suppress the debug messages by giving or changing thi
- parameter to a lower value. debug=0 means no debug messsages
- As default, it's set to the verbose level 2
- Since this option can be changed dynamically via sysfs file, you ca
- suppress the verbose debug messages on the fly, which wasn't possibl
- before
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Reword information messages for BIOS auto-probing mod
- The sentense "Unknown model for xxx, ..." makes people too nervou
- and drives them to a direction to a wrong "fix" by giving an
- mismatching model option
- Let's rephrase the messages to be more nice and easy (at least tha
- won't make people suspect conspiracies)
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add description of new models for ALC889/889
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: pcm - Add logging of hwptr updates and interrupt update
- Added the logging functionality to xrun_debug to record the hwpt
- updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt()
- corresponding to 16 and 8, respectively
- For example
- # echo 9 > /proc/asound/card0/pcm0p/xrun_debu
- will record the position and other parameters at each period interrup
- together with the normal XRUN debugging
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Merge patch_alc882() and patch_alc883(
- Merge patch_alc882() and patch_alc883() to the former one since bot
- codecs have fairly similar connections but just a slight difference
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - More description about patch module optio
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add description about patch loadin
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix support for Samsung P50 with AD1986A code
- Samsung P50 requires the HP auto-muting unlike other Samsung models
- Added a new model=samsung-p50 to support this
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add model=6530g optio
- Add the new model string corresponding to the previous Acer Aspir
- 6530G support
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - trivial: Miscellaneous documentation typo fixe
- Fix various typos in documentation txts
- Signed-off-by: Matt LaPlante <kernel1@cyberdogtech.com
- Signed-off-by: Jiri Kosina <jkosina@suse.cz
- - ALSA: pcm - Update document about xrun_debug proc fil
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add 7.1 support for MSI GX62
- Added 7.1 support for MSI GX620 and jack quirk
- Reference: kernel bug#1343
- http://bugzilla.kernel.org/show_bug.cgi?id=1343
- Signed-off-by: David Heidelberger <d.okias@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: support Sony Vaio T
- with BIOS probing only we offer a non functional headphone swith an
- volume slider
- Signed-off-by: Guido Günther <agx@sigxcpu.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ice1724 - Add ESI Maya44 suppor
- Added the support for ESI Maya44 board to ice1724 driver
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Acer Aspire 8930G suppor
- Short story: this laptop has 5.1 built-in speakers which you *really
- want to use (the not-so-"sub" woofer is what makes the audio abov
- average for a laptop), so 6-channel support is important (plus a decen
- asound.conf to upmix stereo). It also has the 3 typical jacks that ough
- to have a selectable mode. And it's based on ALC889, which sucks
- Rationale/explanations
- The const_channel_count stuff was added because, for a laptop like this
- you always have 6 channels available (internal speakers) but still nee
- to set the mode for the 3 external jacks. Therefore, the device alway
- needs to be in 6-channel mode but there still needs to be a mixe
- control for the jack mode. You could use line/mic-in at the same time a
- the 6 internal speakers, for example. You might be tempted to make i
- even smarter by dynamically switching the max channel count whe
- headphones are plugged in (therefore muting the internal speakers an
- reducing the physical channel count to the jack channel mode), but as
- user I consider this to be harmful because I want the audio to blow u
- to 6 channels / upmixed as soon as I unplug the headphones, and havin
- opened the device while in 2-channel mode would prevent this fro
- working (and always making 6-channel mode available doesn't do any harm)
- The hardware needs EAPD turned on and the DACs routed to the interna
- speaker pins, so the patch adds those verbs
- The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT wor
- by default, at least here. I wasted much time trying to talk t
- Realtek/pshou about this, but they just kept sending me useless update
- to patch_realtek.c that did nothing relevant. In the end I gave up an
- brute forced the issue by trying to flip every bit in the proprietar
- coefficient registers, and eventually found the two magic registers tha
- need to be cleared to enable all DACs. I have only heard Acer user
- complain, but that might be because ALC889 is pretty new and using 5.
- (and noticing the missing center/lfe channels) might not be that common
- If this is a generalized issue with all ALC889 systems then those verb
- should probably be moved to a common verb array
- The internal mic is untested and probably doesn't work
- These settings will probably work for other Acer Gemstone laptops wit
- the same 5.1 speaker config. When identified, those should be added t
- the PCI subsystem ID list
- Signed-off-by: Hector Martin <hector@marcansoft.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Enable PCM hw_ptr_jiffies check only in xrun_debug mod
- The PCM hw_ptr jiffies check results sometimes in problems when
- hardware doesn't give smooth hw_ptr updates. So far, au88x0 and som
- other drivers appear not working due to this strict check
- However, this check is a nice debug tool, and the capability should b
- still kept
- Hence, we disable this check now as default unless the user enables i
- by setting the xrun_debug mode to the specific stream via a proc file
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Improved MacBook 3,1 suppor
- This patch adds support for MacBook 3,1 sound by adding a model ne
- "mb31" with the appropriate init verbs, mixers and channel modes t
- the ALC883 configuration. patch_alc882() and patch_alc883() ar
- modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A
- correctly
- Signed-off-by: Torben Schulz <public@letorbi.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: SB X-Fi driver merg
- The Sound Blaster X-Fi driver supports Creative solutions based o
- 20K1 and 20K2 chipsets
- Supported hardware
- Creative Sound Blaster X-Fi Titanium Fatal1ty® Champion Serie
- Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Serie
- Creative Sound Blaster X-Fi Titanium Professional Audi
- Creative Sound Blaster X-Fi Titaniu
- Creative Sound Blaster X-Fi Elite Pr
- Creative Sound Blaster X-Fi Platinu
- Creative Sound Blaster X-Fi Fatal1t
- Creative Sound Blaster X-Fi XtremeGame
- Creative Sound Blaster X-Fi XtremeMusi
- Current release features
- * ALSA PCM Playbac
- * ALSA Recor
- * ALSA Mixe
- Note
- * External I/O modules detection not included
- Signed-off-by: Wai Yew CHAY <wychay@ctl.creative.com
- Singed-off-by: Ryan RICHARDS <ryan_richards@creativelabs.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add support of Samsung NC10 mini noteboo
- Add specific configuration for Samsung NC10 mini notebook. Interna
- mic/speakers will be correctly muted when plugging in external ones
- Mixer controls are added for speakers, headphones and PC beep
- "Boost" is added for the microphones
- Signed-off-by: Chris Pockelé <chris.pockele.f1@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add missing models for Realtek codec
- Added the missing descriptions and the model names for Realtek codec
- to the documentation and the config table
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: sc6000: enable joystick por
- Add module parameter to enable or disabl
- joystick port (gameport) on the SC6600 an
- later cards
- Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Addition for HP dv4-1222nr laptop suppor
- Signed-off-by: James Gardiner <renidragsemaj@yahoo.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add power supply widget to DAP
- Many modern CODECs have shared resources on chip which must be enable
- for portions of the chip to work but which can be disabled at other time
- in order to achieve power savings. Examples of such resources includ
- power supplies and some internal clocks
- Since these widgets are dependencies for the audio path but do not carr
- audio signals they require slightly different handling to most widgets
- they do not contribute to the audio path and so should not be counted a
- either inputs or outputs during path walks
- Cases where one supply provides a supply for another will requir
- additional work. There is also room for more optimisation of the grap
- walking to avoid repeated checks for the same thing
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Add missing description of lx6464es to ALSA-Configuration.tx
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add 5stack-no-fp model for STAC927
- The recent fix for the headphone volume control on IDT/STAC codec
- resulted in the removal of invalid "Side" volume eventually. But
- if the front panel doesn't exist, this setup could be regarded as
- sort of regression, as reported in kernel bug #13250
- Now as a workaround, a new model 5stack-no-fp is added so that the use
- without the front panel can choose this one explicitly
- Reference: bko#1325
- http://bugzilla.kernel.org/show_bug.cgi?id=1325
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - sound: virtuoso: add Xonar Essence ST suppor
- Add support for the Asus Xonar Essence ST and its daughterboard
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
EMU10K1/EMU10K2 driver
- - Remove multiple KERN_ prefixes from printk format
- Commit 5fd29d6ccbc98884569d6f3105aeca70858b3e0f ("printk: clean u
- handling of log-levels and newlines") changed printk semantics. print
- lines with multiple KERN_<level> prefixes are no longer emitted a
- before the patch
- <level> is now included in the output on each additional use
- Remove all uses of multiple KERN_<level>s in formats
- Signed-off-by: Joe Perches <joe@perches.com
- Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org
- - sound: Use PCI_VDEVICE for CREATIVE and ECTIV
- Here's a patch on top of the others to use CREATIVE and ECTIV
- Signed-off-by: Joe Perches <joe@perches.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: emu10k1 - Fix minimum periods for efx playbac
- EFX playback stream should have periods_min = 2 to avoid the buffe
- position overflow (due to restrictions of the pcm-indirect helper)
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: Remove invalid GENERIC_MIX PCM sublas
- SNDRV_PCM_SUBCLASS_GENERIC_MIX is mostly for h/w multi-stream playbac
- devices, but ca0106 and emu10k1x don't support it (unlike emu10k1)
- We shouldn't set that flag to avoid confusion
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: clean up the logic for building sequencer module
- Instead of mangling the CONFIG_* variables in the makefiles over an
- over, set a few helper variables in Kconfig
- Signed-off-by: Michal Marek <mmarek@suse.cz
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
ENS1370/1+ driver
- - sound: Use PCI_VDEVICE for CREATIVE and ECTIV
- Here's a patch on top of the others to use CREATIVE and ECTIV
- Signed-off-by: Joe Perches <joe@perches.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
ES1688 driver
- - ALSA: Add missing __devexit_p() marker
- 3 ISA sound drivers lack their __devexit_p() markers, which woul
- cause build failures when the kernel is built without hotplug support
- Signed-off-by: Jean Delvare <khali@linux-fr.org
- Cc: Kyle McMartin <kyle@mcmartin.ca
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Echoaudio driver
- - ALSA: indigo-express: add missing 64KHz flag
- Indigo-express cards also support 64KHz sampling rate: this patch add
- missing SNDRV_PCM_RATE_64000 flags
- Signed-off-by: Giuliano Pochini <pochini@shiny.it
- Signed-off-by: Takashi Iwai <tiwai@suse.de
Emagic Audiowerk 2
- - trivial: typo (en|dis|avail|remove)bale -> (en|dis|avail|remove)abl
- Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com
- Signed-off-by: Jiri Kosina <jkosina@suse.cz
GUS Extreme driver
- - ALSA: Add missing __devexit_p() marker
- 3 ISA sound drivers lack their __devexit_p() markers, which woul
- cause build failures when the kernel is built without hotplug support
- Signed-off-by: Jean Delvare <khali@linux-fr.org
- Cc: Kyle McMartin <kyle@mcmartin.ca
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
GUS Library
- - ALSA: sound/isa: convert nested spin_lock_irqsave to spin_loc
- If spin_lock_irqsave is called twice in a row with the same secon
- argument, the interrupt state at the point of the second call overwrite
- the value saved by the first call. Indeed, the second call does not nee
- to save the interrupt state, so it is changed to a simple spin_lock
- The semantic match that finds this problem is as follows
- (http://www.emn.fr/x-info/coccinelle/
- // <smpl
- @
- expression lock1,lock2
- expression flags
- @
- *spin_lock_irqsave(lock1,flags
- ... when != flag
- *spin_lock_irqsave(lock2,flags
- // </smpl
- Signed-off-by: Julia Lawall <julia@diku.dk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
Generic drivers
- - time: move PIT_TICK_RATE to linux/timex.
- PIT_TICK_RATE is currently defined in four architectures, but in thre
- different places. While linux/timex.h is not the perfect place for it, i
- is still a reasonable replacement for those drivers that traditionally us
- asm/timex.h to get CLOCK_TICK_RATE and expect it to be the PIT frequency
- Note that for Alpha, the actual value changed from 1193182UL to 1193180UL
- This is unlikely to make a difference, and probably can only improv
- accuracy. There was a discussion on the correct value of CLOCK_TICK_RAT
- a few years ago, after which every existing instance was getting change
- to 1193182. According to the specification, it should b
- 1193181.818181..
- Signed-off-by: Arnd Bergmann <arnd@arndb.de
- Cc: Richard Henderson <rth@twiddle.net
- Cc: Ivan Kokshaysky <ink@jurassic.park.msu.ru
- Cc: Ralf Baechle <ralf@linux-mips.org
- Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org
- Cc: Ingo Molnar <mingo@elte.hu
- Cc: Thomas Gleixner <tglx@linutronix.de
- Cc: "H. Peter Anvin" <hpa@zytor.com
- Cc: Len Brown <lenb@kernel.org
- Cc: john stultz <johnstul@us.ibm.com
- Cc: Dmitry Torokhov <dtor@mail.ru
- Cc: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Andrew Morton <akpm@linux-foundation.org
- Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org
- - ALSA: pcsp - fix printk format warning agai
- The commit 5a641bcd6398841cc4606b0a732d41a09256fd94 changed th
- printk format to '%lu', but the value passed seems to be dependen
- on the architecture. On x86-64, I got a new warning now because a
- int value is passed actaully
- As a workaround, just cast the value always to unsigned long
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: pcsp: fix printk format warnin
- Fix printk format warning
- sound/drivers/pcsp/pcsp_mixer.c:54: warning: format '%d' expects type 'int', but argument 3 has type 'long unsigned int
- Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
HDA Codec driver
- - Add build stub for pci/hda/patch_cirrus.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix probe of Toshiba laptops with ALC268 code
- There are many variants of Toshiba laptops with ALC268 codec, an
- it seems that a few of them don't work with model=toshiba prese
- since they have the secondary ALC268 codec just for HDMI output
- This is a regression due to the previous clean-up work to merge al
- Toshiba quirk entries into a single check
- This patch adds the identification of such laptops to apply th
- standard BIOS-probing method. Unfortunately, Toshiba laptops hav
- all the same PCI SSID, so we need to check the codec SSID to identif
- each device
- Tested-by: Alexey Dobriyan <adobriyan@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Enable HP output with Macbook Pro 5,
- The patch below, to be applied on the latest sound-unstable-2.6.git
- enables headphones output on my MacBookPro 5,5, together with th
- automuting feature
- Here is the exact soundcard id
- Vendor Id: 0x1013420
- Subsystem Id: 0x106b4d0
- Revision Id: 0x10030
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - don't build digital output controls if not exis
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix compile warnings in patch_cirrus.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix the speaker volume control nam
- Increase the name string buffer size so that "Surround Speaker Playbac
- Volume" won't be truncated
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add GPIO setup for MacBook pro 5,5 with CS420
- GPIO3 seems corresponding to EAPD that is required for the speake
- output
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add quirk for MacBook Pro 5,5 with CS420
- Add the default pin configs for MBP55
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix double creation of SPDIF input control
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add CS420x-specific coef setu
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Force to initialize input mixer setup for CS420
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix cirrus codec parsin
- The parser wasn't called in the proper order
- Split now the parser to be called in patch_cirrus(), and the res
- are just for building PCMs and controls
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add more quirk for HP laptops with AD1984
- More entries for HP laptops to get them working properly
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add full audio support on Acer Aspire 7730G noteboo
- 1) Added support of internal subwoofer (it sounds!!!
- 2) Auto muting front speakers and internal subwoofer on headphones plug
- 3) Internal mic works
- 4) 3 channel mods (jack maps)
- black pink blu
- 2ch: front ext mic line i
- 4ch: front ext mic surroun
- 6ch: front CLFE surroun
- Can be changed in mixer
- 5) Sound can be recorded from
- Internal mi
- Ext mi
- C
- Line i
- 6) 2 separate capture channels
- Signed-off-by: Denis Kuplyakov <dener.kup@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Improve auto-cfg mixer name for ALC66
- The last patch in this series is for ALC662; pretty similar as th
- previous patch for ALC861-VD
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Improve auto-cfg mixer name for ALC861-V
- One more patch to give a better name for the primary output controls
- this time for ALC861-VD codec. The change is simple, just checking th
- pin connection whether it's a speaker-out. When both speaker and H
- are assigned, we name the volume as "PCM" as this influences on bot
- outputs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Improve auto-cfg mixer name for ALC26
- Similar improvements for ALC262 codec like previous two commits
- assign a better name, either Master or Speaker, for the primary outpu
- controls
- However, in the case of ALC262 codec, the necessary changes are large
- than others because we need to check the possibility of different mixe
- amps depending on the pins. The pin 0x16 is mono, and bound with th
- dedicated mixer 0x0e while other pins are bound with 0x0c. Thus, ther
- are two possible volumes
- When only one of them is used, we can name it as "Master". OTOH, whe
- both are used at the same time, they have to be named uniquely
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Improve auto-cfg mixer name for ALC26
- Instead of fixed "Front" mixer name, try to assign a better name, e.g
- "Master" or "Speaker" fot the primary output volume controls of ALC26
- codec
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Improve auto-cfg mixer name for ALC88
- When there is only one DAC is used for ALC880, try to assign a bette
- name, either Speaker or Front, depending on the output pin type
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Generalize input pin parsing in patch_realtek.
- Provide a standard parser for input pins to create the input mixe
- and input source controls instead of having a difference one for eac
- Realtek codec. The new helper parses the codec connections dynamicall
- isntead of fixed indicies
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Reuse ALC268 parser for ALC26
- Reuse a part of the code of ALC268 parser for ALC269
- This will change the default output volume either to Front or Speake
- depending on the pin configuration
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda: move open coded tricks into get_wcaps_channels(
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix invalid capture mixers with some ALC268 model
- The auto-mic clean-up patches caused regressions on some ALC268 model
- that have no proper input_mux but with "Input Source" mixer elements
- Such a combination results in Oops when accessed
- [A reason why set_capture_mixer() isn't used in patch_alc268() is tha
- ALC268 codec have HDA_OUTPUT direction for capture volumes unlike othe
- codecs. Thus it needs own definitions of capture elements.
- This patch fixes the issues
- - Add a capture mixer definition without input-sourc
- - Use the new capture mixer appropriatel
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add missing num_adc_nids definition for IDT92HD8xx
- The previous fix removed the definition of num_adc_nids wrongly, an
- this resulted in the missing input-source control. Now readded again
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix / clean up IDT92HD83xxx codec parse
- A few improvements for IDT 92HD83xxx codec pareser
- - Remove unused / deprecated mixer-amp control
- - Handle d-mics as normal inputs since this codec has no separat
- MUXes for analog and digita
- - Don't create duplicated controls for capture volumes with Mu
- capture volume
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Enable line-out detection only with speaker
- Enable line-out detection for IDT/STAC codecs only when speaker pin
- exist. In some cases, the speaker itself is identified as line-out
- and this confuses the situation. Only the extra line-outs should d
- auto-muting
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - fix noise issue when recording from digital mic with alc26
- With auto config model of alc268 realtek codec, it allows to select an
- of possible available digital microphone inputs when only one i
- available. For example, when only digital mic in nid 0x12 is available
- on second input source it will allow you to select unavailable digita
- mic in nid 0x13. The problem is that selecting unavailable digital mi
- creates a source of noise when recording (I'm not sure if this happen
- on all machines with alc268 and only one digital mic input, but testin
- on a quanta uw1 netbook a lot of noise is introduced in recording fro
- digital mic 0x12/first input source, when you select the unavailabl
- digital mic 0x13 for capture source 0x24 in the second input source i
- mixer)
- Then to avoid noise when recording from digital mic with auto model i
- this case, prevent a digital mic input source to be selected i
- microphone is not available
- Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Clean up init and setup hooks for Realtek codec
- Move static codes to setup from init_hook for each model
- Also, use the common auto-mic selection helper for devices that suppor
- auto-mic selection. They just need to set up ext_mic, int_mic an
- auto_mic flag in the setup section
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add setup hook to ALC preset struc
- Added setup hook to ALC preset struct to be called at in the parse
- but not at each init callback
- This can be used for setting up the static pins, etc, while th
- init hook should be used for updating the status again
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Check connectivity for auto-mic of Realtek codec
- Some Realtek codecs don't provide the full connections for certain pin
- from each ADC; e.g. ACL662/ALC272 gives only one of two digital-mic pin
- for each ADC. Thus, depending on the digital mic pin, the ADC/MUX to b
- used has to be chosen properly
- This patch adds the check of the connectivity of pins at auto-mic mode
- If no proper connectivity is found, auto_mic flag is turned off to b
- sure
- Also the mux_idx is determined during this check so it won't be checke
- in the unsol event any more
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Use only one capture stream for auto-mi
- When the auto-mic feature is enabled, we should support only on
- capture stream
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add auto-mic support for Realtek codec
- Added the support for automatic mic selection via plugging fo
- Realtek codecs (in auto-probing mode). The auto-mic mode is enable
- only when one internal mic and one external mic are present
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix Oops due to STAC/IDT auto-mic change
- The previous auto-mic patch for STAC/IDT codecs causes the Oops o
- machines without digital mic pins. This patch fixes the problem
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add quirks for some HP laptop
- The new HP laptops have PCI SSID 103c:701x and requires model=hp-dv5
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix line-out jack handling with STAC/IDT code
- When the line-out jack is plugged/unplugged, the driver needs to chec
- the headphone plug, not only the line-out jack itself. Otherwise th
- headphone or the speaker may be wrongly muted/unmuted
- As a result, both STAC_HP_EVENT and STAC_LO_EVENT need to call th
- same function, stac92xx_hp_detect()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix line-out jack detectio
- The commit fefd67f31ee7f5259344e36a237d59b47e8715c
- ALSA: hda - Add line-out jack detection on IDT/STAC codec
- enabled wrong pins for jack detections. Fixed to the correct ones
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda: add IbexPeak/Clarkdale HDMI model with static cvt/pin numbe
- The new IbexPeak HDMI codec has 3 pin nodes and 2 converter nodes
- Here we assume only the first ones will be used
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add line-out jack detection on IDT/STAC codec
- Add the automatic mute of speakers via line-out jack plugging o
- STAC/IDT codecs. The feature is enabled when the HP detect is present
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Integrate Digital Input Source to Input Sourc
- STAC/IDT codecs provide both "Input Source" and "Digital Input Source
- controls to choose the analog input source and the digital input source
- But this is far user-unfriendly
- This patch merges the input source selections into one "Input Source
- control. To have separate digital and analog input source controls
- you can pass "separate_dmux = 1 " hint string
- At the same time, this patch gets rid of analog mixer stuff that wa
- already disabled in previous patches
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add Cirrus Logic CS420x suppor
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda: add model for Intel DG45ID/DG45FC board
- The BIOS pin configs are in fact correct and shall not be overwritten
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda: enable speaker output for Compaq 6530s/6531
- HP Compaq 6530s and 6531s internal speaker is silence or becomes silenc
- within 1 minute after fresh boot. It is found that pin 0x1c must be set t
- PIN_OUT mode to make the speaker work. This is weird - line-in pin 0x1c an
- speaker pin 0x16 seem to be unrelated
- The codec differences before/after patch are
- @@ Node 0x17 [Pin Complex] wcaps 0x40020b
- Pin Default 0x41a6e130: [N/A] Mic at Ext Rea
- Conn = Digital, Color = Whit
- DefAssociation = 0x3, Sequence = 0x
- Misc = NO_PRESENC
- - Pin-ctls: 0x24: I
- + Pin-ctls: 0x40: OU
- @@ Node 0x1c [Pin Complex] wcaps 0x40018d
- Pin Default 0x41813021: [N/A] Line In at Ext Rea
- Conn = 1/8, Color = Blu
- DefAssociation = 0x2, Sequence = 0x
- - Pin-ctls: 0x24: IN VREF_8
- + Pin-ctls: 0x40: OUT VREF_HI
- Unsolicited: tag=00, enabled=
- Connection:
- 0x2
- Tests show that it won't impact (external) Mic recording
- Reported-by: "Lin, Ming M" <ming.m.lin@intel.com
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Don't override ADC definitions for ALC codec
- ALC269 and ALC861-VD parsers override the ADC definition
- unconditionally without checking the spec definition. This cause
- the problem when any inconsistent ADC is set up in the device quir
- (like ALC272 with digital-mic)
- This patch avoids the overriding by adding the proper checks
- Reference: Novell bnc#52946
- https://bugzilla.novell.com/show_bug.cgi?id=52946
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add missing vmaster initialization for ALC26
- Without the initialization of vmaster NID, the dB information go
- confused for ALC269 codec
- Reference: Novell bnc#52736
- https://bugzilla.novell.com/show_bug.cgi?id=52736
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Cc: <stable@kernel.org
- - ALSA: hda - Read buffer overflo
- Check whether index is within bounds before testing the element
- Signed-off-by: Roel Kluin <roel.kluin@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda: Correct EAPD for Dell Inspiron 152
- The commit 24918b61b55c21e09a3e07cd82e1b3a8154782dc statically change
- the model from dell-bios to dell-3stack to solve the sound decreasin
- regression (http://lkml.org/lkml/2008/9/12/203), however it leads to anothe
- problem that the 2nd headphone jack doesn't wor
- (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I thin
- the commit 249**2dc is just a workaround. I would like to give a true solutio
- here
- The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, an
- the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD a
- GPIO2. This patch changes EAPD to GPIO0 to solve the problem
- Signed-off-by: Chengu Wang <wangchengu@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda: track CIRB/CORB command/response states for each code
- Recently we hit a bug in our dev board, whose HDMI codec#3 may emi
- redundant/spurious responses, which were then taken as responses t
- command for another onboard Realtek codec#2, and mess up both codecs
- Extend the azx_rb.cmds and azx_rb.res to array and track each codec'
- commands/responses separately. This helps keep good codec safe fro
- broken ones
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix quirk for Toshiba Satellite A135-S452
- Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S452
- with ALC861-VD codec
- Reference: Novell bnc#52632
- https://bugzilla.novell.com/show_bug.cgi?id=52632
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Increase PCM stream name buf in patch_realtek.
- The name buf with size 16 is too short for some codec names, e.g
- truncated like "ALC861-VD Analo". Now the size is doubled
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix typos of Capture controls
- The commit 6479c63188290beae83ade3243b9d6eb47d394b
- ALSA: hda - Create Capture controls dynamicall
- introduced typos of "Capture". Fixed now
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda: add HP automute support to Intel ALC889/ALC889A model
- It auto mutes all 8-channel outputs at rear panel whe
- the front panel headphone is connected
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda: add 2-channel mode to Intel ALC889/ALC889A model
- This 2-channel mode is useful in that it will broadcas
- a 2-channel audio stream to all front/side/... ports
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - No analog mix input source as default for IDT92HD71bx
- The analog mix is disabled now as default (unless "analog_mixer" hin
- is given), so it shoudn't appear in the digital input source as well
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add missing DMUX initialization for auto-mic with STAC/ID
- Added the missing initialization of DMUX connection (to analog input
- for auto-mic mode with STAC/IDT codecs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Remove static connection in IDT 92HD71bx
- We don't need any more static connection to the port F (which is ofte
- used for docking stations) since its connection is done dynamically vi
- DAC assignment now
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Support auto-mic switching with IDT/STAC code
- Support the automatic mic-switching with some devices with IDT/STA
- codecs. The condition is that the device has only two inputs, on
- for an external mic and one for an internal mic
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Avoid overwrite of jack events with STAC/ID
- Since only one event can be associated to a (pin) widget, it's safe
- to avoid the multiple mapping. This patch fixes the behavior of th
- STAC/IDT codec driver
- Now stac_get_event() doesn't take the type argument but simply return
- the first hit element. Then enable_pin_detect() checks the validit
- of the type, and returns non-zero only if a valid entry. The calle
- can call stac_issue_unsol_event() after checking the return value
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Don't create analog mixer for IDT92HD71bx
- The analog mixer unit on IDT 92HD71Bxx codecs is almost useles
- since we use only the direct connections from DAC to pin
- Remove the controls to avoid unneeded confusion as default now
- This can be still back via "analog_mixer = 1" hint
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Create Capture controls dynamicall
- Instead of static snd_kcontrol_new arrays, create "Capture Volume
- and "Capture Switch" controls dynamically based on the mixer att
- values (made via HDA_COMPOSE_AMP_VAL())
- This reduces the code size and gives more flexibility to chang
- the number of controls later
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Don't create unneeded digital input source for IDT 92HD71
- The current driver creates always the digital input source mixe
- elements for IDT 92HD71x codecs no matter whether digital mics ar
- present. This patch adds the proper check to avoid the creation o
- these controls if unnecessary
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Reword information messages for BIOS auto-probing mod
- The sentense "Unknown model for xxx, ..." makes people too nervou
- and drives them to a direction to a wrong "fix" by giving an
- mismatching model option
- Let's rephrase the messages to be more nice and easy (at least tha
- won't make people suspect conspiracies)
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add quirk for Dell Studio 155
- Added a quirk entry for Dell Studio 1555
- Reference: Novell bnc#52524
- https://bugzilla.novell.com/show_bug.cgi?id=52524
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add exception for volume-knob in snd_hda_get_connections(
- Volume-knob widgets may have connections even if they have no CONN_LIS
- cap bit. Allow the query exceptionally in snd_hda_get_connections()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Introduce get_wcaps_type() macr
- Add a helper macro to retrieve the widget type from wiget cap bits
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix mute control with some ALC262 model
- The master mute switch is wrongly implemented as checking the pointe
- instead of its value, thus it can be never muted. This patch fixe
- the issue
- Reference: Novell bnc#40487
- https://bugzilla.novell.com/show_bug.cgi?id=40487
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Cc: <stable@kernel.org
- - [ALSA] Add better Intel IbexPeak platform suppor
- Here are the new sound enabling patches for IbexPeak
- Summary of tested features
- - playbac
- - Front Headphone: O
- - 8 channel audio: Front/Rear/CLFE/Side all O
- - recordin
- - Front Mic/Rear Mic: both O
- (front/rear/line mics are selectable in the "Input source" alsamixer control
- - Line In: not workin
- (in 6ch mode, its amp/mute, direction and route all looks fine
- so I'm a little puzzled
- (hopefully no one will care this feature
- - digital SPDIF input/output: not tested (no equipment
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Restore GPIO1 properly at resume with AD1984
- The commit 099db17e66294b02814dee01c81d9abbbeece93e introduced
- regression at suspend/resume where the GPIO1 bit isn't properl
- restored, thus the speaker output gets muted initially after resume
- The fix is simple, use the cached write for storing GPIO data
- Reference: Novell bnc#52276
- https://bugzilla.novell.com/show_bug.cgi?id=52276
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Use snprintf() to be safe
- Use snprint() for creating the jack name string instead of sprintf(
- in patch_sigmatel.c
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix ALC861 auto-mode parse
- Fix the logic of ALC861 auto-mode parser for the outputs
- Instead of assuming the fixed DAC list, parse the conection and assig
- the DAC dynamically
- Also, unmute the unused output connections to avoid noises on inputs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Reduce click noise at power-savin
- Add some tricks to reduce the click noise at powering down to D
- in the power saving mode on STAC/IDT codecs
- The key seems to be to reset PINs before the power-down, and som
- delay before entering D3. The needed delay is significantly long
- but I don't know why
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix pin-setup for Sony VAIO with STAC9872 codec
- The recent rewrite of the codec parser for STAC9872 caused a regressio
- for some Sony VAIO models that don't give proper pin default config
- by BIOS. Even using model=vaio doesn't work because the pin definition
- are set after the pin overrides
- This patch fixes the pin definitions in patch_stac9872() to be pu
- in the right place before the pin overrides. Also the patch adds th
- new quirk entry for VAIO F/S to have the correct pin default configs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Cc: <stable@kernel.org
- - ALSA: hda - Add quirk for Gateway T6834c lapto
- Gateway T6834c laptops need EAPD always on while the default behavio
- for the STAC9205 reference board is to turn it off upon every HP plug
- By using the special "eapd" model, which is first introduced for Gatewa
- T1616 laptops for this same reason, this peculiarity can be properl
- handled
- Signed-off-by: Hao Song <baritono.tux@gmail.com
- Cc: <stable@kernel.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - [ALSA] hda-intel: Cleanups for widget connection list handlin
- This patch adds a check to snd_hda_get_connections() routine fo
- presence of AC_WCAP_CONN_LIST. Also, make sure that negative erro
- codes from noted route are handled on all places as errors
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - [ALSA] hda_codec: Check for invalid zero connection
- To prevent "Too many connections" message and the error path for some HDM
- codecs (which makes onboard audio unusable), check for invalid zer
- connections for CONNECT_LIST verb
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix ALC268 parser for mono speake
- - Parse the mono output pin 0x16 correctly even as the primary outpu
- - Create "Speaker" volume control if the primary output is a speake
- - Fix the wrong direction of (optional) "Mono" switc
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix the previous sanity check in make_codec_cmd(
- The newly added sanity-check for a codec verb can be better writte
- with logical ORs. Also, the parameter can be more than 8bit
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - add bounds checking for the codec command field
- A recent bug involves passing auto detected >0x7f NID to codec command
- creating an invalid codec addr field, and finally lead to cmd timeou
- and fall back into single command mode. Jaroslav fixed that bug i
- alc880_parse_auto_config()
- It would be safer to further check the bounds of all cmd fields
- Cc: Jaroslav Kysela <perex@perex.cz
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add CX20582 and OLPC XO-1.5 suppor
- This adds support for the Conexant CX20582 codec, based on code fro
- http://www.linuxant.com/alsa-driver/alsa-driver-linuxant-1.0.19ppch12-1.noarch.rpm.zi
- This is the codec to be shipped in the OLPC XO-1.5, so this patch als
- includes an XO-specific profile. Resultant configuration
- http://dev.laptop.org/~dsd/20090713/codec0.tx
- http://dev.laptop.org/~dsd/20090713/codec0.sv
- As the Linuxant code is structured differently to the other codecs
- I was unable to cleanly reimplement everything in the generic and Del
- profiles as more info is needed (e.g. codec graphs). I simplified thos
- profiles so that hopefully it will not break anyone's audio. If it does
- it may be worth returning -ENODEV from patch_cx5066 on non-OLPC systems
- and then fixing snd_hda_codec_configure() to fall back on the generi
- parser, at least until support for other systems is figured out
- Signed-off-by: Daniel Drake <dsd@laptop.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Check codec errors in snd_hda_get_connections(
- The codec read errors in snd_hda_get_connections() are ignored so far
- and it causes a problem like the bug in the commi
- 9d30937accf2c01e8b0bd59787409a7348cbbcb
- ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checkin
- Better to check errors in the function and returns a proper error cod
- rather than passing bogus NID values
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix the merge erro
- Fix the merge error at the commit 305355aad89f1b7eb27cb210fad2f9d3c67b2572
- an addition of the missing alc880_gpio3_init_verbs to ALC882_TARGA model
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checkin
- On some IbexPeak systems with ALC889A errors like "azx_get_respons
- timeout, switching to polling mode: last cmd=0xaf9f000b" are produced
- because non-existent codec #10 is wrongly accessed
- The problem is that snd_hda_get_connections() returns out-of-range resul
- for NID 0x1c (something like 0xf8f9 or 0xffff)
- This patch adds a check to alc880_parse_auto_config() to avoid usin
- of this out-of-range NIDs. A better fix maybe to improv
- snd_hda_get_connections() routine to check for valid NID ranges i
- NIDs are expected as result
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - targa and targa-2ch fi
- Simplify ALC882_TARGA and return gpio3 to ALC883_TARGA_DIG an
- ALC883_TARGA_2ch_DIG, which I accidentally removed in commit i
- 64a8be74357477558183b43156c5536b642de13
- Signed-off-by: David Heidelberger <d.okias@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Missing volume controls for Intel HDA (ALC269/EeePC
- There is a regression, introduced in aa202455eec51699e44f658530728162cefa130
- (in alsa-kernel) which I noticed when trying to use the headphone socket o
- my EeeCPC 901: the output was *very* quiet, practically silent
- This patch corrects the control types to that which was obviously intended i
- the referenced commit
- Signed-off-by: Darren Salt <linux@youmustbejoking.demon.co.uk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add quirks for RTL888 & RV630/M76 based MSI GX71
- Signed-off-by: William Weston <weston@sysex.net
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Check widget types while parsing capture source in patch_via.
- Check the widget type and don't take invalid widgets while parsin
- the capture source in patch_via.c
- Also, fixed some compile warnings introduced in the previous commit
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix capture source selection in patch_via.
- The fixed widget NIDs in patch_via.c seem wrong for some codecs
- and it resulted in the invalid capture source selection
- This patch adds the code to parse the topology instead of usin
- fixed numbers in order to get the right MUX widget id correspondin
- to the ADCs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add missing EAPD initialization for VIA codec
- If the output pin is used and EAPD capability is present, turn o
- the EAPD bit. This fixes the silent output problem on ASUS laptop
- with VT1708S codec
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Clean up VT170x dig-in initialization cod
- Minor clean up for initializing the digital-in pin
- No functional changes
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - move 8086:fb30 quirk (stac9205) to the proper sectio
- Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Don't override maxbps for FLOAT sharing with linear format
- When FLOAT PCM format is available but together with other linea
- PCM formats, don't override maxbps value. For FLOAT format, it's alway
- 32, thus it can be better checked in snd_hda_calc_stream_format()
- Otherwise the maxbps 32 might be used wrongly even if the linear PC
- doesn't support it
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Manually expand alc882_init_verb
- Instead of expanding alc882_init_verbs to two elements via a macro
- manually expand to each entry. This makes clear that some have alread
- the full slot for init_verbs array (currently 5)
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add missing mixer amp initialization for ALC88
- After merting patch_alc882() and patch_alc883(), the initialization o
- mixer amp 0x0b was missing in alc882_base_init_verbs[]
- This is usually no critical problem, but it can disable the power-savin
- as the default state, so better to put to mute these channels
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Allow FLOAT PCM forma
- So far, the FLOAT PCM format is used only exclusivley set. Bu
- this can be a combination with other formats
- This patch changes the parser to allow the FLOAT format in additio
- to other PCM formats
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix input pinctl for ALC882 auto mod
- alc882_auto_init_analog_input() sets the input pins to VREF-80 regardles
- of the input pin types although it shouldn't be for line-in pins
- This patch fixes the behavior to follow other codecs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Merge patch_alc882() and patch_alc883(
- Merge patch_alc882() and patch_alc883() to the former one since bot
- codecs have fairly similar connections but just a slight difference
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add patch module optio
- Added the patch module option to apply a "patch" as a firmware t
- modify pin configurations or give additional hints to the drive
- before actually initializing and configuring the codec
- This can be used as a workaround when the BIOS doesn't give sufficien
- information or give wrong information that doesn't match with the rea
- hardware setup, until it's fixed statically in the driver via a quirk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Don't call snd_hda_codec_configure in snd_hda_codec_new(
- The codec setup call via snd_hda_codec_configure() isn't necessaril
- called in snd_hda_codec_new(). For the later added feature, it's bette
- to change the code flow like
- - create all codec instance
- - configure each code
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Avoid invalid formats and rates with shared SPDI
- Check whether formats and rates don't result in zero due to th
- restriction of SPDIF sharing. If any of them can be zero, disabl
- the SPDIF sharing mode instead. Otherwise it will lead to a PC
- configuration error
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Improve ASUS eeePC 1000 mixe
- The mixer elements created for ASUS eeePC 1000 with ALC269 aren'
- standard but strange words like "LineOut". Rename the element name
- to follow the standard one like "Headphone" and "Speaker"
- Also, split the volumes to each so that the virtual master can contro
- them
- The alc269_fujitsu_mixer is removed because it's now identical wit
- the new eeepc mixer
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add GPIO1 control at muting with HP laptop
- HP laptops with AD1984A codecs (at least mobile models) need to se
- GPIO1 appropriately to indicate the mute state. The BIOS checks thi
- bit to judge whether the mute on or off is sent via F8 key
- Without changing this bit, the BIOS can be confused and may toggl
- the mute wrongly
- Reference: Novell bnc#51526
- https://bugzilla.novell.com/show_bug.cgi?id=51526
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add quirk for HP 6930
- Added a quirk model=laptop for HP 6930p (103c:30dc) with AD1984A codec
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add missing static to patch_ca0110(
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add missing initializations for ALC268 and ALC26
- During the changes to clean up / fix the realtek codec initializatio
- routines in commit 4a79ba34cada6a5a4ee86ed53aa8a73ba1e6fc51
- I forgot to add the check for ALC268 and ALC269
- This resulted in the missing EAPD and COEF setup for these codecs
- This patch adds the missing checks for these codecs
- Reference: bko#1363
- http://bugzilla.kernel.org/show_bug.cgi?id=1363
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Line In for Acer Inspire 6530G mode
- The Line In connector is set up as PIN_IN by default, usin
- VREF_HIZ. It is connected to both ADCs, so add it to bot
- input selectors
- Also add the ability to use the input mix (on a SoundBlaste
- one would call this "What You Hear")
- Signed-off-by: Tony Vroon <tony@linx.net
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Use model=acer-aspire-6530g for Acer Aspire 6930
- For Acer Aspire 6930G (1025:015e), acre-aspire-6530g model matche
- obviously better
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix acer-aspire-6530g model quir
- Fix the following bugs of acer-aspire-6530g model with ALC888
- - HP jack to mute all speaker outputs including LF
- - Make digital built-in mic workin
- Signed-off-by: Emilio López <buhitoescolar@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add pin-sense trigger when needed for Realtek codec
- Realtek codecs require the pin-sense trigger call before actuall
- reading the pin-sense. Without this, the pin-detection might not b
- done accurately
- This patch adds the pin-capability check and issues the trigger cal
- if required
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix support for Samsung P50 with AD1986A code
- Samsung P50 requires the HP auto-muting unlike other Samsung models
- Added a new model=samsung-p50 to support this
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Generalize the pin-detect quirk for Lenovo N10
- Add a new flag to ad_spec struct so that the same hack can be used fo
- any other models (if any). This also allows other models to reuse th
- auto-mute functions
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Simplify AD1986A mixer definition
- Split mixer element arrays of AD1986A models to several pieces so tha
- each model can share the same mixer arrays
- This removes lots of duplicated data
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Make jack-plug notification selectabl
- Make the jack-plug notification via input layer selectable via Kconfig
- This is often unnecessary, and the similr function will be provide
- using the ALSA control API in near future anyway
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add digital-mic support to ALC262 auto mode
- Add the digital-mic support with ALC262 auto model
- The new ALC262 models have the digital mic at NID 0x12. This widge
- isn't checked in the current alc262_auto_create_analog_input_ctls(
- since it's under 0x18. So, just reuse the routine for alc269 to fi
- the behavior
- But, it doesn't suffice: the digital mic is supported only with th
- ADC0, we have to exclude other ADCs when d-mic is detected
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix check of input source type for realtek codec
- Fix the check of the input-source type by checking the widget type o
- each capture-source item. Since some codecs can have both the mixe
- and selector types depending on the ADC, alc_mux_enum_put() needs t
- check each widget
- With this change, spec->capture_style gets unneeded, so it's removed
- too
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add quirk for Sony VAIO Z21M
- It needs model=toshiba-s06 to work with the digital-mic
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Cc: <stable@kernel.org
- - ALSA: hda - Get back Input Source for ALC262 toshiba-s06 mode
- The commit f9e336f65b666b8f1764d17e9b7c21c90748a37
- ALSA: hda - Unify capture mixer creation in realtek code
- removed the "Input Source" mixer element creation for toshiba-s06 mode
- because it contains a digital-mic input
- This patch take the control back
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Cc: <stable@kernel.org
- - ALSA: hda - Fix unsigned comparison in patch_sigmatel.
- Fix the comparison of unsigned int that causes a compile warning belo
- by changing to the right signed type
- patch_sigmatel.c: In function ‘stac92xx_vref_set’
- patch_sigmatel.c:658: warning: comparison of unsigned expression < 0 is always fals
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add model=6530g optio
- Add the new model string corresponding to the previous Acer Aspir
- 6530G support
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Acer Inspire 6530G model for Realtek ALC88
- The selected 4930G model seemed to keep the subwoofer 'tuba
- function from operating correctly. Removing the existing PC
- ID match made this work again, but it was mapped to 'Side
- instead of to LFE as one would expect
- This attempts to enable all functionality and keep the amoun
- of available mixer sliders low. Any slider that had no audibl
- effect on the output audio has been removed, and as such EAP
- is not currently enabled
- Signed-off-by: Tony Vroon <tony@linx.net
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: HDA - Correct trivial typos in comments
- Correct some trivial typos in comments
- Signed-off-by: Sasha Alexandr <brina_keith@ns.sympatico.ca
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: HDA - Name-fixes in code (tagra/targa
- Correct some cut+paste typos from 'tagra' to 'targa'
- Signed-off-by: Sasha Alexandr <brina_keith@ns.sympatico.ca
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: HDA - Add pci-quirk for MSI MS-7350 motherboard
- Add pci-quirk for MSI MS-7350 motherboard with Realtek ALC888
- Signed-off-by: Sasha Alexandr <brina_keith@ns.sympatico.ca
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix memory leak at codec creatio
- The codec->modelname field is allocated twice in snd_hda_codec_new()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add quirk for Acer Aspire 6935
- Added model=acer-aspire-8930g for Acer Aspire 6935G (1025:0146)
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - add quirk for STAC92xx (SigmaTel STAC9205
- A quirk is required for 8086:284b (rev 03) [Subsystem: 161f:2073]
- The following has been tested with Alsa 1.0.20 (git master)
- Background details can be found a
- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=456
- http://forum.ubuntu-gr.org/viewtopic.php?f=38&t=529
- Tested-by: Theodora Iliopoulou <th30dr@gmail.com
- Signed-off-by: Simos Xenitellis <simos@gnome.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix the previous tagra-8ch patc
- - Fix a typo in the patc
- - Adapted to follow the recent change for unsol event handlin
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add 7.1 support for MSI GX62
- Added 7.1 support for MSI GX620 and jack quirk
- Reference: kernel bug#1343
- http://bugzilla.kernel.org/show_bug.cgi?id=1343
- Signed-off-by: David Heidelberger <d.okias@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: support Sony Vaio T
- with BIOS probing only we offer a non functional headphone swith an
- volume slider
- Signed-off-by: Guido Günther <agx@sigxcpu.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - More Aspire 8930G fixe
- Enable all three capture channels, including the missing nid 7 which i
- the only one capable of capturing DMIC inpu
- Enable Headphone amp for the HP jack. This causes a volume boost fo
- headphones, but does not cause any noticeable effect for light load
- like other amps, so there is no need to make it configurable
- Add Input Mix capture mux setting to capture the output of the playbac
- input mux (that is, what goes out the speakers except for PCM
- Hack another coef register because the stereo DMIC for some reaso
- produces a nonstandard sum/difference signal. I found a bit to make i
- just use the sum signal for both channels, which makes it behave like
- standard mono microphone. The stereo is useless anyway (they're 1cm apart)
- Tested working: Three capture channels, mic in, line in, DMIC
- Tested not working: CD. Not sure why, might be unconnected in the actua
- hardware or a CD drive issue
- Also looked at SPDIF. It appears to work (emitter lights up inside th
- HP out jack) but I lack a proper miniTOSLINK cable to test it
- Signed-off-by: Hector Martin <hector@marcansoft.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Limit codec-verb retry to limited hardware
- The reset of a BUS controller during operations is somehow risky an
- shouldn't be done inevitably for devices that have apparently no suc
- codec-communication problems
- This patch adds the check of the hardware and limits the bus-rese
- capability
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add codec bus reset and verb-retry at critical error
- Some machines machine cause a severe CORB/RIRB stall in certai
- weird conditions, such as PA access at the start up together wit
- fglrx driver. This seems unable to be recovered without the controlle
- reset
- This patch allows the bus controller reset at critical errors s
- that the communication gets recovered again
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Acer Aspire 8930G suppor
- Short story: this laptop has 5.1 built-in speakers which you *really
- want to use (the not-so-"sub" woofer is what makes the audio abov
- average for a laptop), so 6-channel support is important (plus a decen
- asound.conf to upmix stereo). It also has the 3 typical jacks that ough
- to have a selectable mode. And it's based on ALC889, which sucks
- Rationale/explanations
- The const_channel_count stuff was added because, for a laptop like this
- you always have 6 channels available (internal speakers) but still nee
- to set the mode for the 3 external jacks. Therefore, the device alway
- needs to be in 6-channel mode but there still needs to be a mixe
- control for the jack mode. You could use line/mic-in at the same time a
- the 6 internal speakers, for example. You might be tempted to make i
- even smarter by dynamically switching the max channel count whe
- headphones are plugged in (therefore muting the internal speakers an
- reducing the physical channel count to the jack channel mode), but as
- user I consider this to be harmful because I want the audio to blow u
- to 6 channels / upmixed as soon as I unplug the headphones, and havin
- opened the device while in 2-channel mode would prevent this fro
- working (and always making 6-channel mode available doesn't do any harm)
- The hardware needs EAPD turned on and the DACs routed to the interna
- speaker pins, so the patch adds those verbs
- The ALC889 CLFE and subsequent (side/aux, here unused) DACs do NOT wor
- by default, at least here. I wasted much time trying to talk t
- Realtek/pshou about this, but they just kept sending me useless update
- to patch_realtek.c that did nothing relevant. In the end I gave up an
- brute forced the issue by trying to flip every bit in the proprietar
- coefficient registers, and eventually found the two magic registers tha
- need to be cleared to enable all DACs. I have only heard Acer user
- complain, but that might be because ALC889 is pretty new and using 5.
- (and noticing the missing center/lfe channels) might not be that common
- If this is a generalized issue with all ALC889 systems then those verb
- should probably be moved to a common verb array
- The internal mic is untested and probably doesn't work
- These settings will probably work for other Acer Gemstone laptops wit
- the same 5.1 speaker config. When identified, those should be added t
- the PCI subsystem ID list
- Signed-off-by: Hector Martin <hector@marcansoft.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Reorder and clean-up ALC268 quirk tabl
- Rearrange alc268_cfg_tbl[] in the order of vendor id, and group som
- entries using SND_PCI_QUIRK_MASK()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - fix audio on LG R51
- Currently, LG R510 is only able to produce sound on headphones, th
- internal speakers are not working
- The user tested and confirmed that with model=Dell headphones
- internal speakers and the microphone are working flawlessly
- Tested-by: Serdar Soytetir <tulliana@gmail.com
- Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Macbook[Pro] 5 6ch suppor
- this is a patch against current snapshot that adds
- 6 channels support for the MB5 mode
- Signed-off-by: Kacper Szczesniak <kacper@qwe.pl
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Jack Mode changes for Sigmatel board
- This patch changes Line In as Out Switch and Mic In as Out Switch t
- enums for consistency, and causes all mic and line in ports to be probe
- and controls to be added appropriately
- Signed-off-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Support NVIDIA 8 channel HDMI audi
- Support 8 channel HDMI audio for MCP78/7
- Signed-off-by: Wei Ni <wni@nvidia.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda-intel: improve initialization for ALC262_HP_BPC mode
- Fix issues for 3 generations of HP workstations
- The modest modifications do the following
- 1. Change the second MIC from device 3 to device
- 2. Init the "boost" values to "0" by defaul
- From: John Brown <john.brown3@hp.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix reverted LED setup for H
- The commit 86d190e77c44cb057742dcc871b12ebd4633c387 reverted the bi
- flip of LED GPIO for HP DX and DV4-1222nr. Fixed now
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Use snd_hda_codec_get_pincfg() in patch_ca0110.
- Use the new function to reduce the access and allow the user setu
- via sysfs, too
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix channels_max setting for CA011
- Added the missing definition of max channels for CA0110, which resulte
- in an error at opening PCM devices
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Minor clean up of patch_sigmatel.
- - Remove unneeded semicolon
- - Introduce spec->gpio_led to specify the GPIO bit for LED contro
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Compaq Presario CQ60 patching for Conexan
- A docking mic control is shown by default. The Compaq Presari
- CQ60 laptop has no docking connector, so designate it as
- CXT5051_HP model
- This makes the phantom mixer slider disappear
- Signed-off-by: Tony Vroon <tony@linx.net
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Support sync after writing a ver
- This patch adds a debug mode to make the codec communicatio
- synchronous. Define SND_HDA_SUPPORT_SYNC_WRITE in hda_codec.c
- and the call of snd_hda_codec_write*() will become synchronous
- i.e. wait for the reply from the codec at each time issuing a verb
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix digital beep tone calculatio
- The digital beep tone is calculated in two different ways dependin
- on the codec chip. The standard one is using a divider, and anothe
- one is a linear tone for IDT/STAC codecs. Currently, only th
- latter type is used for all codecs, which resulted in a wrong ton
- pitch
- This patch adds the calculation of the standard HD-audio type
- Also clean-up the fields in hda_beep struct
- Reference: bko#1316
- http://bugzilla.kernel.org/show_bug.cgi?id=1316
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Improved MacBook 3,1 suppor
- This patch adds support for MacBook 3,1 sound by adding a model ne
- "mb31" with the appropriate init verbs, mixers and channel modes t
- the ALC883 configuration. patch_alc882() and patch_alc883() ar
- modified to handle the MacBook 3,1 sound-chip (Realtek ALC889A
- correctly
- Signed-off-by: Torben Schulz <public@letorbi.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Show the actual chip name in 'unkown model' message
- Show the actual chip name in 'unknown model..' info messages fo
- Realtek codecs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Split codec->name to vendor and chip name string
- Split the name string in hda_codec struct to vendor_name and chip_nam
- strings to be stored directly from the preset name
- Since mostly only the chip name is referred in many patch_*.c, thi
- results in the reduction of many codes in the end
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - add controls to toggle DC bias on mic port
- This patch adds a mixer control for the STAC92XX boards to control th
- DC bias of mic ports, allowing recording from both powered an
- non-powered sources. It replaces the "Mic Output Switch" with "Mic Jac
- Mode" to switch between Mic, Line In, and Line Out
- Signed-off-by: Nickolas Lloyd <ultrageek.lloyd@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add a quirk entry for Macbook Pro 5,
- Added the codec SSID for MacBook Pro 5,1 as compatible as MP51
- However, the headphone auto-muting function doesn't work. So
- this is just a tentative solution
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Disable fallback to model=acer for Acer laptop
- The model=acer for ALC883/889 doesn't work well for the recent Ace
- Aspire laptops. Since model=auto works better nowadays, it's safe
- to use the default fallback instead of the Acer specific one
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add support of Samsung NC10 mini noteboo
- Add specific configuration for Samsung NC10 mini notebook. Interna
- mic/speakers will be correctly muted when plugging in external ones
- Mixer controls are added for speakers, headphones and PC beep
- "Boost" is added for the microphones
- Signed-off-by: Chris Pockelé <chris.pockele.f1@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add missing models for Realtek codec
- Added the missing descriptions and the model names for Realtek codec
- to the documentation and the config table
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Clean up Realtek auto-mute unsol routine
- Most of unsol handlers defined in patch_realtek.c can be classified t
- two types, mute via amp of pins and mute via ctl bits of pins
- Thus there are a big room to generalize each implementation
- This patch creates two generic functions, alc_automute_amp() an
- alc_automute_pin(). The latter is actually changed from the previou
- alc_sku_automute(). Each caller needs to initialize hp_pins an
- speaker_pins properly at own init_hook
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Clean up for ALC262 HP model auto-mute function
- Just clean up, no functional changes
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix and clean up hippo-compat HP auto-mutin
- The speaker auto-muting per HP plugging for ALC262 HIPPO and compatibl
- devices is slightly buggy as the "Master" or "Front" mixer control ca
- still toggle the speaker output even if the headphone is plugged
- This patch fixes the issue, and clean up the hippo-related code
- together with fixes of some inconsistent mixer names
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix secondary SPDIF on VT1708S and VT1702 codec
- VIA VT1708S and VT1702 codecs can have two SPDIF outputs. One of the
- should have been handled as the extra digital out, but it's no
- properly accessed
- This patch fixes the handling of the secondary SPDIF on these codec
- with the slave dig-out as found in patch_sigmatel.c. This makes th
- use of such a device easier (for normal users)
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add support for MacBook 5.1 (Aluminium
- Signed-off-by: Kacper Szczesniak <kacper@qwe.pl
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Addition for HP dv4-1222nr laptop suppor
- Signed-off-by: James Gardiner <renidragsemaj@yahoo.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix a typo in patch_realtek.c agai
- The commmit dfed0ef9b3ff9e37903920b6938ed33344ad0b3d was reverte
- accidentally by the merge of auto-detection fix patch
- Fixed again now
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Don't enable auto-mute but for speakers in patch_realtek.
- Enable auto-muting in model=auto only for devices with HP and speakers
- For devices with HP and line-outs, don't enable the auto-muting
- Also, add a debug print to show the auto-mute feature
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add amp initialization for realtek auto mod
- In the realtek auto-probing mode, the initialization of amp wit
- some magic COEF or EAPD verbs is applied only when the codec SSI
- has valid values to satisfy the realtek's definition
- However, many devices don't provide in that way, thus the devic
- doesn't work as is
- This patch allows the same initialization code even if the SSI
- doesn't pass the bit test. Also, alc_subsystem_id() is change
- just to check and define the type, so that it's called in th
- parser, instead of the initializer
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix a typo in debug print for realtek auto-detectio
- The NID and ASS numbers were swapped..
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - minor optimization in hda_set_power_state(
- Check the target power-state before checking EAPD exception to reduc
- unneeded verb executions
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add debug prints for Realtek auto-ini
- Added a couple of debug prints to show the checked id numbers i
- alc_subsystem_id()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Retry codec-verbs at error
- The current error-recovery scheme for the codec communication error
- doesn't work always well. Especially falling back to th
- single-command mode causes the fatal problem on many systems
- In this patch, the problematic verb is re-issued again after the erro
- (even with polling mode) instead of the single-cmd mode. Th
- single-cmd mode will be used only when specified via the comman
- option explicitly, mainly just for testing
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Cache PCM and STREAM parameters querie
- Cache quries for PCM and STREAM parameters as well as ampcap an
- pincap sharing the hash table. This will reduce the superfluou
- access of the same codec verbs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Check strcpy lengt
- Check the length to copy via strlen() beforehand to avoid the stac
- corruption, or use strlcpy() to be safe in HD-audio codes
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add Creative CA0110-IBG suppor
- Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode
- In the HD-audio mode, no multiple streams are supported by just i
- behaves like a normal HD-audio device
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add missing check of pin vref 50 and others in Realtek codec
- Some Realtek codecs like ALC861 seem to support only VREF50 while th
- current driver assumes it's only VREF80. Check other VREF bits to se
- the correct value
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add 5stack-no-fp model for STAC927
- The recent fix for the headphone volume control on IDT/STAC codec
- resulted in the removal of invalid "Side" volume eventually. But
- if the front panel doesn't exist, this setup could be regarded as
- sort of regression, as reported in kernel bug #13250
- Now as a workaround, a new model 5stack-no-fp is added so that the use
- without the front panel can choose this one explicitly
- Reference: bko#1325
- http://bugzilla.kernel.org/show_bug.cgi?id=1325
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - fix audio on HP TX25xx series notebook
- Fixes https://bugtrack.alsa-project.org/alsa-bug/view.php?id=412
- Taken from https://bugzilla.redhat.com/show_bug.cgi?id=49806
- Signed-off-by: Adam Williamson <awilliam@redhat.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix line-in on Mac Mini Core2 Du
- BIOS on Mac Mini Core2 Duo sets both INPUT and OUTPUT pinctl bits t
- the line-in jack, and it confuses the driver as if it's a valid input
- This patch adds the check of OUTPUT bit so that the driver fixes th
- invalid pin setup
- Tested-by: Tino Keitel <tino.keitel@gmx.de
- Cc: <stable@kernel.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
HDA Intel driver
- - Fix build of hda_intel.
- The commit dc4c2e6bde77735071dbef7aca6bd6c0116102b3 in sound tre
- causes the build errors on older kernels due to undefined PCI id an
- the use of pci_dev.revirsion field. Make a patch to fix the build
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add a white-list for MSI optio
- Created a white-list to enable MSI since some devices require MS
- explicitly due to BIOS/ACPI problems. Simply using a quirk list
- As the first case, take HP Compaq CQ40
- Reference: Novell bnc#52997
- https://bugzilla.novell.com/show_bug.cgi?id=52997
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda: warn on spurious respons
- To help disclose hardware bugs
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda: remember last command for each code
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda: read CORBWP inside reg_loc
- This converts the last CORBWP access outside of reg_lock
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_i
- Just for safety. azx_init_cmd_io() and azx_free_cmd_io() may b
- called when switching to single command mode
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda: take cmd_mutex in probe_codec(
- Now that each codec will have its own module, it is possibl
- for the user to load one codec while another one is running
- So cmd_mutex would be a safe addition to probe_codec()
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda: track CIRB/CORB command/response states for each code
- Recently we hit a bug in our dev board, whose HDMI codec#3 may emi
- redundant/spurious responses, which were then taken as responses t
- command for another onboard Realtek codec#2, and mess up both codecs
- Extend the azx_rb.cmds and azx_rb.res to array and track each codec'
- commands/responses separately. This helps keep good codec safe fro
- broken ones
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Add support for new AMD HD audio device
- Add support for new AMD HD audio devices. Use generic driver to detect HD audi
- devices with Vendor ID AMD
- Signed-off-by: Andiry Xu <andiry.xu@amd.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Disable AMD SB600 64bit address support onl
- HDA driver disabled HD audio 64bit address support for all AM
- SB600/SB700/SB800 platforms with commi
- 09240cf429505891d6123ce14a29f58f2a60121e due to one SB600 issu
- reported by community, but we do not see the similar issue o
- SB700/SB800 platforms
- This patch is to refine the workaround for SB600 only
- Signed-off-by: Andiry Xu <andiry.xu@amd.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix error path in the sanity check in azx_pcm_open(
- Release resources cleanly after errors in the sanity check i
- azx_pcm_open()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add patch module optio
- Added the patch module option to apply a "patch" as a firmware t
- modify pin configurations or give additional hints to the drive
- before actually initializing and configuring the codec
- This can be used as a workaround when the BIOS doesn't give sufficien
- information or give wrong information that doesn't match with the rea
- hardware setup, until it's fixed statically in the driver via a quirk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Don't call snd_hda_codec_configure in snd_hda_codec_new(
- The codec setup call via snd_hda_codec_configure() isn't necessaril
- called in snd_hda_codec_new(). For the later added feature, it's bette
- to change the code flow like
- - create all codec instance
- - configure each code
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add sanity check in PCM open callbac
- Add some sanity checks of struct snd_pcm_hardware fields in the PC
- open callback of hda driver. This makes a bit easier to debug any PC
- setup errors in the codec side
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Call snd_pcm_lib_hw_rates() again after codec open callbac
- The PCM rates bit field may have been changed by the codec open callback
- In that case, we need to reset rate_min and rate_max. So, simply cal
- snd_pcm_lib_hw_rates() again after the codec open callback
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda_intel: fix build error when !P
- Fix this build error when CONFIG_PM is not set
- ound/pci/hda/hda_intel.c: In function 'azx_bus_reset'
- sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all
- sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend
- sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume
- Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Limit codec-verb retry to limited hardware
- The reset of a BUS controller during operations is somehow risky an
- shouldn't be done inevitably for devices that have apparently no suc
- codec-communication problems
- This patch adds the check of the hardware and limits the bus-rese
- capability
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add codec bus reset and verb-retry at critical error
- Some machines machine cause a severe CORB/RIRB stall in certai
- weird conditions, such as PA access at the start up together wit
- fglrx driver. This seems unable to be recovered without the controlle
- reset
- This patch allows the bus controller reset at critical errors s
- that the communication gets recovered again
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Fix a typo in the previous patc
- ICH6_GCTL_RESET was wrongly set to another bit by the commi
- b21fadb9c1852c91622ca1dccfeb144bc535e36e. This caused a problem whe
- the codec needs really a reset (e.g. recovering from the communicatio
- error at probe)
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add more register bits definition
- Added some missing register bits definitions to reduce magic numbers
- Also renamed some to follow the names on the datasheet
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Always sync writes in single_cmd mod
- In the single_cmd mode, the hardware cannot store the multiple replie
- like on RIRB, thus each verb has to sync and wait for the response n
- matter whether the return value is needed or not. Otherwise it ma
- result in a wrong return value from the previous verb
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Allow concurrent RIRB access in single_cmd mod
- In the single_cmd mode, the current driver code doesn't do any updat
- for RIRB just for any safety reason. But, actually the RIRB an
- single_cmd mode don't conflict. Unsolicited events can be delivere
- even while using the single_cmd mode
- This patch allows the handling of unsolicited events with single_cm
- mode, just always checking RIRB independent from single_cmd flag
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Reset CORB/RIRB at retrying the verb communicatio
- When a codec communication error occurs, the CORB/RIRB counters shoul
- be reset first before re-issuing the verb. Simply call azx_free_cmd_io(
- and azx_init_cmd_io() to achieve that
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add prefix to kernel message
- Add proper prefix to each kernel message in hda_intel.c
- Also, avoid the unneeded prefix when CONFIG_SND_VERBOSE_PRINTK is use
- together with snd_print*()
- Reference: bko#1320
- http://bugzilla.kernel.org/show_bug.cgi?id=1320
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Avoid conflicts with snd-ctxfi drive
- The PCI entries of Creative with HD-audio class can be the device
- with emu20k1/emu20k2 chips. These are supported better by snd-ctxf
- driver. With that driver, the device will mutate from HD-audio t
- its native class
- This patch adds a simple ifdef to avoid the conflict of device prob
- between snd-hda-intel and snd-ctxfi drivers. 1102:0009 seems stil
- OK to be added as it has no emu20kx chip, and is a pure HD-audi
- device
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Retry codec-verbs at error
- The current error-recovery scheme for the codec communication error
- doesn't work always well. Especially falling back to th
- single-command mode causes the fatal problem on many systems
- In this patch, the problematic verb is re-issued again after the erro
- (even with polling mode) instead of the single-cmd mode. Th
- single-cmd mode will be used only when specified via the comman
- option explicitly, mainly just for testing
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Check strcpy lengt
- Check the length to copy via strlen() beforehand to avoid the stac
- corruption, or use strlcpy() to be safe in HD-audio codes
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add Creative CA0110-IBG suppor
- Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode
- In the HD-audio mode, no multiple streams are supported by just i
- behaves like a normal HD-audio device
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add forced codec-slots for ASUS W5F
- ASUS W5Fm needs the fixed codec-slots to probe to override the BIO
- problem like W5F
- Tested-by: Alp Kılıç <kilic.alp@gmail.com
- Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr
- Signed-off-by: Takashi Iwai <tiwai@suse.de
HDA generic driver
- - Fix build of hda_intel.
- The commit dc4c2e6bde77735071dbef7aca6bd6c0116102b3 in sound tre
- causes the build errors on older kernels due to undefined PCI id an
- the use of pci_dev.revirsion field. Make a patch to fix the build
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda: move open coded tricks into get_wcaps_channels(
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add Cirrus Logic CS420x suppor
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda: fix out-of-bound hdmi_eld.sad[] writ
- e->sad[] is declared with size ELD_MAX_SAD=16, but the guar
- allows range 0-31
- Signed-off-by: Roel Kluin <roel.kluin@gmail.com
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Introduce get_wcaps_type() macr
- Add a helper macro to retrieve the widget type from wiget cap bits
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - [ALSA] hda_generic: use AC_WCAP_CONN_LIST check for widget connection
- Previous patch used widget type, but the presence flag of the connectio
- list is in the widget capabilities
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - [ALSA] hda_generic: do not read connections for widged with an unknown typ
- Reading node connections for an unknown widget can confuse HDA codec bus
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - fix beep tone calculation for IDT/STAC codec
- In the beep tone calculation for IDT/STAC codecs, lower numbers correspon
- to higher frequencies and vice versa. The current code has this backwards
- resulting in beep frequencies which are way too high (and sound bad o
- tinny laptop speakers, resulting in complaints)
- [Also added hz <= 0 check by tiwai
- Signed-off-by: Paul Vojta <vojta@math.berkeley.edu
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Check "beep" hin
- Check the hint "beep" in snd_hda_attach_beep_device() to avoid the bee
- device creation if user doesn't want
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add patch module optio
- Added the patch module option to apply a "patch" as a firmware t
- modify pin configurations or give additional hints to the drive
- before actually initializing and configuring the codec
- This can be used as a workaround when the BIOS doesn't give sufficien
- information or give wrong information that doesn't match with the rea
- hardware setup, until it's fixed statically in the driver via a quirk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Don't call snd_hda_codec_configure in snd_hda_codec_new(
- The codec setup call via snd_hda_codec_configure() isn't necessaril
- called in snd_hda_codec_new(). For the later added feature, it's bette
- to change the code flow like
- - create all codec instance
- - configure each code
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Make jack-plug notification selectabl
- Make the jack-plug notification via input layer selectable via Kconfig
- This is often unnecessary, and the similr function will be provide
- using the ALSA control API in near future anyway
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: hda - Fix digital beep tone calculatio
- The digital beep tone is calculated in two different ways dependin
- on the codec chip. The standard one is using a divider, and anothe
- one is a linear tone for IDT/STAC codecs. Currently, only th
- latter type is used for all codecs, which resulted in a wrong ton
- pitch
- This patch adds the calculation of the standard HD-audio type
- Also clean-up the fields in hda_beep struct
- Reference: bko#1316
- http://bugzilla.kernel.org/show_bug.cgi?id=1316
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Split codec->name to vendor and chip name string
- Split the name string in hda_codec struct to vendor_name and chip_nam
- strings to be stored directly from the preset name
- Since mostly only the chip name is referred in many patch_*.c, thi
- results in the reduction of many codes in the end
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hda - Add Creative CA0110-IBG suppor
- Added the support for Creative SB X-Fi boards with UAA (HD-audio) mode
- In the HD-audio mode, no multiple streams are supported by just i
- behaves like a normal HD-audio device
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
I2C UDA1380
- - ASoC: UDA1380: refactor device registratio
- This patch mostly follows commit 5998102b9095fdb7c67755812038612afea315c
- "ASoC: Refactor WM8731 device registration" to make UDA1380 use standar
- device instantiation. Similarly, the I2C device registration temporaril
- moves into the magician machine driver before it will find its fina
- resting place in the board file
- At the same time, platform specific configuration is moved to platform dat
- and common power/reset GPIO handling moves into the codec driver
- Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
ICE1712 driver
- - Add build stub for ice1724 maya44 suppor
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD
- I've built a small HTPC and had to add suspend/resume support in ice172
- driver. There seem to be 3 existing bugs related to that
- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=441
- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=374
- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=231
- Due to hardware (un)availability, I only enabled the fix for Audiotra
- Prodigy HD2 card, which is installed in my HTPC. However, most of my cod
- should be reusable in the future on other ice1724-based cards as well (a
- long as people add card-specific peices of code). The fix is currently base
- on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel)
- Signed-off-by: Igor Chernyshev <igor.ch75+alsa at gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ice1724 - Add ESI Maya44 suppor
- Added the support for ESI Maya44 board to ice1724 driver
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ice1724 - Allow spec driver to create own routing control
- Added a new flag, own_routing, to allow spec drivers to create ow
- routing controls. Also, the basic get/put calls are changed to b
- external for later use by maya44 driver
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
ICE1724 driver
- - ALSA: ice1724 - Fix section mismatc
- Now snd_vt1724_chip_reset() is used in the resume callback, thu
- it cannot be __devinit
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ice1724 - Patch for suspend/resume for Audiotrak Prodigy HD
- I've built a small HTPC and had to add suspend/resume support in ice172
- driver. There seem to be 3 existing bugs related to that
- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=441
- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=374
- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=231
- Due to hardware (un)availability, I only enabled the fix for Audiotra
- Prodigy HD2 card, which is installed in my HTPC. However, most of my cod
- should be reusable in the future on other ice1724-based cards as well (a
- long as people add card-specific peices of code). The fix is currently base
- on ALSA 1.0.20 and works on my MythBuntu 9.04 HTPC (using 2.6.28-11 kernel)
- Signed-off-by: Igor Chernyshev <igor.ch75+alsa at gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ice1724 - Add ESI Maya44 suppor
- Added the support for ESI Maya44 board to ice1724 driver
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ice1724 - Allow spec driver to create own routing control
- Added a new flag, own_routing, to allow spec drivers to create ow
- routing controls. Also, the basic get/put calls are changed to b
- external for later use by maya44 driver
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ice1724 - Add PCI postint to reset sequenc
- Add the PCI posting to ensure the reset sequence in snd_vt1724_chip_reset()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ice1724 - Clean up definitions of DMA record
- Rename some vt1724_pcm_reg records to more generic and consistent ones
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ice1724 - Check error in set_rate functio
- The set_rate might return error but the current code doesn't check it
- This patch adds a proper error check
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
ISA
- - ALSA: sc6000: add support for SC-6600 and SC-700
- Add support for later cards based on CompuMedia ASC-9408 chipsets
- These cards were produced by Gallant
- This patch make the OSS aedsp16 driver redundant
- Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Intel8x0 driver
- - ALSA: intel8x0 - Fix PCM position crazines
- The PCM pointer callback sometimes returns invalid positions and thi
- screws up the hw_ptr updater in PCM core. Especially since now th
- jiffies check is optional with xrun_debug, the invalid position i
- handled as is, and causes serious sound skips, etc
- This patch simplifies the position-fix strategy in intel8x0 to be mor
- robust
- - just falls back to the last position if bogus position is detecte
- - another sanity check for the backward move of the position due t
- a race of register update and the base-index updat
- This patch is applicable also for 2.6.30
- Tested-by: David Miller <davem@davemloft.net
- Cc: <stable@kernel.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
KORG1212 driver
- - ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some driver
- Added SNDRV_PCM_INFO_BATCH flag to PCM info field of some drivers tha
- really don't give the precise pointer value
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
LX6464ES
- - ALSA: lx6464es - configure ethersound io channel
- as long as the io channel number is not set by the driver, the car
- is not visible from the ethersound networ
- Signed-off-by: Tim Blechmann <tim@klingt.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - convert some DMA_nnBIT_MASK() caller
- We're about to make DMA_nnBIT_MASK() emit `deprecated' warnings. Convert th
- remaining stragglers which are visible to the x86_64 build
- Cc: FUJITA Tomonori <fujita.tomonori@lab.ntt.co.jp
- Cc: James Bottomley <James.Bottomley@HansenPartnership.com
- Cc: Eric Moore <Eric.Moore@lsil.com
- Cc: Takashi Iwai <tiwai@suse.de
- Cc: "David S. Miller" <davem@davemloft.net
- Cc: Alexander Duyck <alexander.h.duyck@intel.com
- Cc: Yi Zou <yi.zou@intel.com
- Signed-off-by: Andrew Morton <akpm@linux-foundation.org
- Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org
- - ALSA: lx6464es - support standard alsa module parameter
- trivial patch to support the alsa module parameters `index', `id
- and `enable
- Signed-off-by: Tim Blechmann <tim@klingt.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: lx6464es - Disable lx_message_send(
- Disable lx_message_send() function temporarily as it's not use
- anywhere
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: lx6464es - Use snd_card_create(
- Use snd_card_create() instead of the obsoleted snd_card_new()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: lx6464es - driver for the digigram lx6464es interfac
- prototype of a driver for the digigram lx6464es 64 channel ethersoun
- interface
- Signed-off-by: Tim Blechmann <tim@klingt.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
MSND driver
- - ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some driver
- Added SNDRV_PCM_INFO_BATCH flag to PCM info field of some drivers tha
- really don't give the precise pointer value
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Memalloc module
- - ALSA: Fix SG-buffer DMA with non-coherent architecture
- Using SG-buffers with dma_alloc_coherent() is often very inefficien
- on non-coherent architectures because a tracking record could b
- allocated in addition for each dma_alloc_coherent() call
- Instead, simply disable SG-buffers but just allocate normal continuou
- buffers on non-supported (currently all but x86) architectures
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
OPL3
- - ALSA: clean up the logic for building sequencer module
- Instead of mangling the CONFIG_* variables in the makefiles over an
- over, set a few helper variables in Kconfig
- Signed-off-by: Michal Marek <mmarek@suse.cz
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
OPL4
- - ALSA: clean up the logic for building sequencer module
- Instead of mangling the CONFIG_* variables in the makefiles over an
- over, set a few helper variables in Kconfig
- Signed-off-by: Michal Marek <mmarek@suse.cz
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
OSS device core
- - sound: make OSS device number claiming optional and schedule its remova
- If any OSS support is enabled, regardless of built-in or module
- sound_core claims full OSS major number (that is, the old 0-25
- region) to trap open attempts and request sound modules using custo
- module aliases. This feature is redundant as chrdev already has suc
- mechanism. This preemptive claiming prevents alternative OS
- implementation
- The custom module aliases are scheduled to be removed and the previou
- patch made soundcore emit the standard chrdev aliases too to hel
- transition
- This patch schedule the feature for removal in a year and makes i
- optional so that developers and distros can try new things in th
- meantime without rebuilding the kernel. The pre-claiming can b
- turned off by using SOUND_OSS_CORE_PRECLAIM and/or kernel paramete
- soundcore.preclaim_oss
- As this allows sound minors to be individually grabbed by other users
- this patch updates sound_insert_unit() such that if registerin
- individual device region fails, it tries the next available slot
- For details on removal plan, please read the entry added by this patc
- in feature-removal-schedule.txt
- Signed-off-by: Tejun Heo <tj@kernel.org
- Cc: Alan Cox <alan@lxorguk.ukuu.org.uk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: request char-major-* module aliases for missing OSS device
- Till now missing OSS devices emitted sound-slot/service-* modul
- alises instead of the standard char-major-* if a missing device numbe
- is opened if soundcore is loaded. The custom module aliases don'
- have any inherent benefit than backward compatibility
- sound-slot/service-* module aliases is scheduled to be removed and t
- help the transition this patch makes soundcore emit the standar
- module alises along with the custom ones
- Signed-off-by: Tejun Heo <tj@kernel.org
- Cc: Alan Cox <alan@lxorguk.ukuu.org.uk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: do not set DEVNAME for OSS device
- Signed-off-by: Kay Sievers <kay.sievers@vrfy.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Driver Core: sound: add nodename for sound driver
- This adds support to the sound core to report the proper device name t
- userspace for their devices
- Signed-off-by: Kay Sievers <kay.sievers@vrfy.org
- Signed-off-by: Jan Blunck <jblunck@suse.de
- Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de
PARISC Harmony driver
- - ALSA: Add missing __devexit_p() marker
- 3 ISA sound drivers lack their __devexit_p() markers, which woul
- cause build failures when the kernel is built without hotplug support
- Signed-off-by: Jean Delvare <khali@linux-fr.org
- Cc: Kyle McMartin <kyle@mcmartin.ca
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: parisc/harmony: fix printk format warnin
- Fix this warning
- sound/parisc/harmony.c:938: warning: format '%lx' expects type 'long unsigned int'
- but argument 2 has type 'resource_size_t
- Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
PCI drivers
- - ALSA: azt3328: fix Kconfig entr
- This driver is about as far from being experimental as it can ever ge
- for an undocumented card, thus create this patch (interestingly it was the onl
- EXPERIMENTAL remaining in the entire Kconfig file)
- Signed-off-by: Andreas Mohr <andi@lisas.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Remove PAGE_SIZE limitatio
- Remove the limitation of PAGE_SIZE to be 4k by defining the ow
- page size and macros for 4k. 8kb page size could be natively supported
- but it's disabled right now for simplicity
- Also, clean up using upper_32_bits() macro
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: ctxfi - Add depends on X8
- The ctxfi driver requires explicitly the 4k page size, and gives
- build error on architectures with non-4k pages
- As a workaround, just add the kconfig dependency on X86, which i
- the only architecture ever tested
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: SB X-Fi driver merg
- The Sound Blaster X-Fi driver supports Creative solutions based o
- 20K1 and 20K2 chipsets
- Supported hardware
- Creative Sound Blaster X-Fi Titanium Fatal1ty® Champion Serie
- Creative Sound Blaster X-Fi Titanium Fatal1ty Professional Serie
- Creative Sound Blaster X-Fi Titanium Professional Audi
- Creative Sound Blaster X-Fi Titaniu
- Creative Sound Blaster X-Fi Elite Pr
- Creative Sound Blaster X-Fi Platinu
- Creative Sound Blaster X-Fi Fatal1t
- Creative Sound Blaster X-Fi XtremeGame
- Creative Sound Blaster X-Fi XtremeMusi
- Current release features
- * ALSA PCM Playbac
- * ALSA Recor
- * ALSA Mixe
- Note
- * External I/O modules detection not included
- Signed-off-by: Wai Yew CHAY <wychay@ctl.creative.com
- Singed-off-by: Ryan RICHARDS <ryan_richards@creativelabs.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hdsp - Add a comment about external firmwares for hds
- When the hdsp driver is built in kernel, the corresponding firmwar
- files have to be built into the kernel as well (otherwise the boo
- will hang up for fairly long time), but there is no way to contro
- it in Kconfig yet, unfortunately. Instead, show a comment for use
- just to make sure
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: lx6464es - driver for the digigram lx6464es interfac
- prototype of a driver for the digigram lx6464es 64 channel ethersoun
- interface
- Signed-off-by: Tim Blechmann <tim@klingt.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: virtuoso: add Xonar Essence ST suppor
- Add support for the Asus Xonar Essence ST and its daughterboard
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
PDAudioCF driver
- - ALSA: Add missing SNDRV_PCM_INFO_BATCH flag to some driver
- Added SNDRV_PCM_INFO_BATCH flag to PCM info field of some drivers tha
- really don't give the precise pointer value
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
PPC AWACS driver
- - ALSA: powermac - Replace the rest of __init
- All __initdata should be __devinitdata as platform device is hotpluggable
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: sound/ppc: update annotations of serveral function
- [I am not sure if this is the correct approach as I don't know if any o
- this actual hardware or drivers are really hot pluggable.
- Gets rid of these build warnings
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x5c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_new(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_new()
- If .snd_pmac_new is only used by .snd_pmac_probe the
- annotate .snd_pmac_new with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x10c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_burgundy_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_burgundy_init()
- If .snd_pmac_burgundy_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_burgundy_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x164): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_daca_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_daca_init()
- If .snd_pmac_daca_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_daca_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1dc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_tumbler_init()
- If .snd_pmac_tumbler_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_tumbler_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1ec): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_post_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_tumbler_post_init()
- If .snd_pmac_tumbler_post_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_tumbler_post_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x28c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_awacs_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_awacs_init()
- If .snd_pmac_awacs_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_awacs_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2bc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_pcm_new(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_pcm_new()
- If .snd_pmac_pcm_new is only used by .snd_pmac_probe the
- annotate .snd_pmac_pcm_new with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2f8): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_attach_beep(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_attach_beep()
- If .snd_pmac_attach_beep is only used by .snd_pmac_probe the
- annotate .snd_pmac_attach_beep with a matching annotation
- Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
PPC Beep
- - ALSA: sound/ppc: update annotations of serveral function
- [I am not sure if this is the correct approach as I don't know if any o
- this actual hardware or drivers are really hot pluggable.
- Gets rid of these build warnings
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x5c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_new(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_new()
- If .snd_pmac_new is only used by .snd_pmac_probe the
- annotate .snd_pmac_new with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x10c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_burgundy_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_burgundy_init()
- If .snd_pmac_burgundy_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_burgundy_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x164): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_daca_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_daca_init()
- If .snd_pmac_daca_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_daca_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1dc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_tumbler_init()
- If .snd_pmac_tumbler_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_tumbler_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1ec): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_post_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_tumbler_post_init()
- If .snd_pmac_tumbler_post_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_tumbler_post_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x28c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_awacs_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_awacs_init()
- If .snd_pmac_awacs_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_awacs_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2bc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_pcm_new(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_pcm_new()
- If .snd_pmac_pcm_new is only used by .snd_pmac_probe the
- annotate .snd_pmac_pcm_new with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2f8): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_attach_beep(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_attach_beep()
- If .snd_pmac_attach_beep is only used by .snd_pmac_probe the
- annotate .snd_pmac_attach_beep with a matching annotation
- Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
PPC Burgundy driver
- - ALSA: burgundy: timeout message is off by one
- Timeout message is off by one
- Signed-off-by: Roel Kluin <roel.kluin@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: powermac - Replace the rest of __init
- All __initdata should be __devinitdata as platform device is hotpluggable
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: sound/ppc: update annotations of serveral function
- [I am not sure if this is the correct approach as I don't know if any o
- this actual hardware or drivers are really hot pluggable.
- Gets rid of these build warnings
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x5c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_new(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_new()
- If .snd_pmac_new is only used by .snd_pmac_probe the
- annotate .snd_pmac_new with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x10c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_burgundy_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_burgundy_init()
- If .snd_pmac_burgundy_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_burgundy_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x164): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_daca_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_daca_init()
- If .snd_pmac_daca_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_daca_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1dc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_tumbler_init()
- If .snd_pmac_tumbler_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_tumbler_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1ec): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_post_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_tumbler_post_init()
- If .snd_pmac_tumbler_post_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_tumbler_post_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x28c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_awacs_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_awacs_init()
- If .snd_pmac_awacs_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_awacs_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2bc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_pcm_new(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_pcm_new()
- If .snd_pmac_pcm_new is only used by .snd_pmac_probe the
- annotate .snd_pmac_pcm_new with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2f8): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_attach_beep(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_attach_beep()
- If .snd_pmac_attach_beep is only used by .snd_pmac_probe the
- annotate .snd_pmac_attach_beep with a matching annotation
- Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
PPC DACA driver
- - ALSA: sound/ppc: update annotations of serveral function
- [I am not sure if this is the correct approach as I don't know if any o
- this actual hardware or drivers are really hot pluggable.
- Gets rid of these build warnings
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x5c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_new(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_new()
- If .snd_pmac_new is only used by .snd_pmac_probe the
- annotate .snd_pmac_new with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x10c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_burgundy_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_burgundy_init()
- If .snd_pmac_burgundy_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_burgundy_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x164): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_daca_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_daca_init()
- If .snd_pmac_daca_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_daca_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1dc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_tumbler_init()
- If .snd_pmac_tumbler_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_tumbler_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1ec): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_post_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_tumbler_post_init()
- If .snd_pmac_tumbler_post_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_tumbler_post_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x28c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_awacs_init(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_awacs_init()
- If .snd_pmac_awacs_init is only used by .snd_pmac_probe the
- annotate .snd_pmac_awacs_init with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2bc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_pcm_new(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_pcm_new()
- If .snd_pmac_pcm_new is only used by .snd_pmac_probe the
- annotate .snd_pmac_pcm_new with a matching annotation
- WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2f8): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_attach_beep(
- The function __devinit .snd_pmac_probe() reference
- a function __init .snd_pmac_attach_beep()
- If .snd_pmac_attach_beep is only used by .snd_pmac_probe the
- annotate .snd_pmac_attach_beep with a matching annotation
- Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
PPC Keywest driver
- - ALSA: keywest: Get rid of useless i2c_device_name() macr
- The i2c_device_name() macro is used only once and doesn't have muc
- value, it hurts redability more than it helps. Get rid of it
- Signed-off-by: Jean Delvare <khali@linux-fr.org
- Cc: Benjamin Herrenschmidt <benh@kernel.crashing.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
PPC PMAC driver
- - ALSA: powermac - Replace the rest of __init
- All __initdata should be __devinitdata as platform device is hotpluggable
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
PPC PS3 driver
- - ALSA: sound/ps3: Correct existing and add missing annotation
- probe functions should be __devini
- Signed-off-by: Geert Uytterhoeven <Geert.Uytterhoeven@sonycom.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: sound/ps3: Restructure driver sourc
- Sort includes, and reorder code so we can kill the forward declaration
- No functional change
- Signed-off-by: Geert Uytterhoeven <Geert.Uytterhoeven@sonycom.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: sound/ps3: Fix checkpatch issue
- Signed-off-by: Geert Uytterhoeven <Geert.Uytterhoeven@sonycom.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
PPC Tumbler driver
- - ALSA: powermac - Replace the rest of __init
- All __initdata should be __devinitdata as platform device is hotpluggable
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
RME HDSP driver
- - ALSA: hdsp - allow proc reporting with disconnected io bo
- the hdsp driver refuses to report any information via the pro
- interface, if the io box is not connected. with this patch, th
- content of the control and status registers is printed before th
- iobox check
- Signed-off-by: Tim Blechmann <tim@klingt.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Clean up 64bit division function
- Replace the house-made div64_32() with the standard div_u64*() functions
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: hdsp: allow firmware loading from inside the kerne
- Allow the use of the FIRMWARE_IN_KERNEL option with hdsp cards an
- in-kernel driver
- Also corrected a typo in the comment
- Signed-off-by: Raphael Doursenaud <rdoursenaud@free.fr
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
RME9652 driver
- - ALSA: Clean up 64bit division function
- Replace the house-made div64_32() with the standard div_u64*() functions
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SB drivers
- - ALSA: clean up the logic for building sequencer module
- Instead of mangling the CONFIG_* variables in the makefiles over an
- over, set a few helper variables in Kconfig
- Signed-off-by: Michal Marek <mmarek@suse.cz
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SC6000 (CompuMedia ASC-9308 + AD1848) driver
- - ALSA: sc6000: enable joystick por
- Add module parameter to enable or disabl
- joystick port (gameport) on the SC6600 an
- later cards
- Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: sc6000: fix older card initializatio
- The last patch to handle newer cards like SC700
- broke initialization of the SC6000. Fix this
- Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: sc6000: add support for SC-6600 and SC-700
- Add support for later cards based on CompuMedia ASC-9408 chipsets
- These cards were produced by Gallant
- This patch make the OSS aedsp16 driver redundant
- Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SGI O2 Audio
- - ALSA: sgio2audio.c: clean up checkin
- vfree() does it's own 'NULL' check,so no need for check befor
- calling it
- Signed-off-by: Figo.zhang <figo1802@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
SIS7019 driver
- - trivial: fix typos s/paramter/parameter/ and s/excute/execute/ in documentation and source comments
- Signed-off-by: Martin Olsson <martin@minimum.se
- Signed-off-by: Jiri Kosina <jkosina@suse.cz
SoC Audio for Freecale i.MX1x i.MX2x CPUs
- - Add soc/imx/* build stu
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: Staticise unexported variable
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove unneeded i.MX dependency on SN
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix review issues in i.MX2x PCM drive
- Signed-off-by: javier Martin <javier.martin@vista-silicon.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: add machine driver for i.mx27_visstrim_m10 boar
- This adds support for i.mx27_visstrim_sm10 board machine driver whic
- uses an i.mx27 processor plus a wm8974 codec
- It has been tested on a visstrim_sm10 board
- Signed-off-by: Javier Martin <javier.martin@vista-silicon.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: add DAI platform ssi driver for MX
- This adds support for DAI platform for the SSI present in MXC platforms
- It currently does not support i.MX3, the only thing necessary to d
- this is to export DMA data for i.MX3 interface which I haven't don
- because I don't have a i.MX3 based board available
- It has been tested on i.MX27 board
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: add DMA platform driver for MX1x and MX2
- This adds support for DMA platform valid for i.MX1 and i.MX2 platforms
- This is not valid for i.MX3 since it doesn't share the same DM
- interface than i.MX1 and i.MX2
- It has been tested on i.MX27 board
- Signed-off-by: Javier Martin <javier.martin@vista-silicon.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Audio for TXx9
- - Add soc/txx9 build stu
- Just a Makefile, no source links yet
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: txx9aclc: dynamically allocate dmaengine devnam
- Use kasprintf to allocate temporary devname string instead of
- fixed size string
- This fixes "FIXME" introduced on removal of BUS_ID_SIZE
- Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Kill BUS_ID_SIZ
- Remove the use of BUS_ID_SIZE from txx9aclc.c, as BUS_ID_SIZE will b
- removed soon later
- Also, use snprintf() instead of sprintf() as a safer operation
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: Add TXx9 AC link controller driver (v3
- This patch adds support for the integrated ACLC of the TXx9 family
- Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Audio for the Atmel AT32/AT91 System-on-Chip
- - Add missing ASoC build stub
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: Configure WM8731 SYSCLK at startup on AT91SAM9G20-E
- The system clock is currently fixed by the driver and this avoid
- the need for us to handle errors with enabling and disabling MCL
- (which was incorrect previously so this fixes bugs in erro
- handling)
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Disable microphone input for AT91SAM9G20-EK by defaul
- As shipped the board does not have inputs but it is relativel
- straightforward to modify the board to hook them up so suppor
- is provided in the driver. When these modifications have no
- been made enabling the microphone stage can cause problems
- Add an ifdef to disable this by default. Don't put it int
- Kconfig since users will have to get their soldering iron
- out to change things
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Use CODEC as clock master on AT91SAM9G20-E
- This simplifies the driver by removing the need to manuall
- configure dividers within the CPU and improve audio performanc
- by ensuring that the optimal phase relationships between th
- clocks in the system are maintained
- Note that currently this means that for playback to work th
- Output Mixer HiFi switch must be enabled since otherwise CODE
- will not generate the DAC clock
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: correct print specifiers for unsigned
- Unsigned variables should use `%u' rather than `%d'
- Signed-off-by: Roel Kluin <roel.kluin@gmail.com
- Signed-off-by: Andrew Morton <akpm@linux-foundation.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: AFEB9260 drive
- ASoC driver for AT91SAM9260-based AFEB9260 boar
- Signed-off-by: Sergey Lapin <slapin@ossfans.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Audio for the Samsung S3C24XX chips
- - ASoC: neo1973_gta02_wm8753: Replace deprecated s3c_gpio calls with gpioli
- With the s3c platform has implementing gpiolib support the s3c_gpio api has bee
- deprecated
- This patch gets rid of all s3c_gpio calls and replaces them by using gpiolib
- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: neo1973_gta02_wm8753: Replace snd_soc_cnew with snd_soc_add_controls
- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix s3c-i2s-v2 buil
- We now need the PCM header to kick the DMA
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add S3C24xx dependencies for Simtec machine
- No point in building them for S3C64xx, certainly no sense in runnin
- into build issues there
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: S3C platform: Fix s3c2410_dma_started() called at improper tim
- s3c24xx dma has the auto reload feature, when the the trnasfer is done
- CURR_TC(DSTAT[19:0], current value of transfer count) reaches 0, and DM
- ACK becomes 1, and then, TC(DCON[19:0]) will be loaded into CURR_TC. S
- the transmission is repeated
- IRQ is issued while auto reload occurs. We change the DISRC an
- DCON[19:0] in the ISR, but at this time, the auto reload has bee
- performed already. The first block is being re-transmitted by the DMA
- So we need rewrite the DISRC and DCON[19:0] for the next bloc
- immediatly after the this block has been started to be transported
- The function s3c2410_dma_started() is for this perpose, which is calle
- in the form of "s3c2410_dma_ctrl(prtd->params->channel
- S3C2410_DMAOP_STARTED);" in s3c24xx_pcm_trigger()
- But it is not correct. DMA transmission won't start until DMA REQ signa
- arrived, it is the time s3c24xx_snd_txctrl(1) or s3c24xx_snd_rxctrl(1
- is called in s3c24xx_i2s_trigger()
- In the current framework, s3c24xx_pcm_trigger() is always called befor
- s3c24xx_pcm_trigger(). So the s3c2410_dma_started() should be called i
- s3c24xx_pcm_trigger() after s3c24xx_snd_txctrl(1) o
- s3c24xx_snd_rxctrl(1) is called in this function
- However, s3c2410_dma_started() is dma related, to call this function w
- should provide the channel number, which is given b
- substream->runtime->private_data->params->channel. The private_dat
- points to a struct s3c24xx_runtime_data object, which is define i
- s3c24xx_pcm.c, so s3c2410_dma_started() can't be called in s3c24xx_i2s.
- Fix this by moving the call to signal the DMA started to the DA
- drivers
- Signed-off-by: Shine Liu <liuxian@redflag-linux.com
- Signed-off-by: Shine Liu <shinel@foxmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Select core DMA when building for S3C64x
- Ensure that the core DMA support is available when building fo
- S3C64xx
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: S3C24XX: Support for Simtec Hermes board
- Add support for the tlv320aic3x CODEC on the Simtec Hermes board
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: S3C24XX: Add audio core and tlv320aic23 for Simtec board
- Add core support for the range of S3C24XX Simtec boards with TLV320AIC2
- CODECs on them. Since there are also boards with similar IIS routing th
- AMP and the configuration code is placed in a core file for re-use wit
- other CODEC bindings
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: S3C24XX : Align the peroid size to the buffer siz
- > Then it's a driver bug. If unaligned period size is allowed, it mean
- > that the irq is really generated in that period, not at the buffe
- > boundary. Otherwise, it must have a proper hw-constraint to align th
- > period size to the buffer size
- This patch will fix the bug metioned in the above mail. Force the peroi
- size to be aligned with the buffer size
- Based and tested on linux-2.6.31-rc6
- Signed-off-by: Shine Liu <shinel@foxmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Reenable S3C64xx I2S suppor
- Joonyoung Shim reports that S3C64xx I2S is working on the NCP boards s
- allow it to be selected in Kconfig
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmciro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix data format configuration for S3C64XX IISv
- The data format configuration for S3C64xx IISv2 was hardcoded for IISMO
- register. This patch changes to the defined values it
- And instead of bits 9 and 10 of IISMOD we should clear bits 13 and 14
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: s3c2443-ac97: convert semaphore to mute
- This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig)
- CC [M] sound/soc/s3c24xx/s3c2443-ac97.
- sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX
- sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaratio
- sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read'
- sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function
- sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only onc
- sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.
- sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write'
- sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function
- Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Existing S3C24xx AC97 drivers should depend on S3C24x
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add Openmoko Neo FreeRunner (GTA02) audio drive
- This driver supports the audio subsystem on the Openmoko Neo FreeRunne
- smartphone, often known by its codename GTA02. The system has a WM875
- connected to a Samsung S3C2442 with an external GPIO controlled speake
- amplifier
- The driver was originally written by Graeme Gregory and has recieve
- contributions from Openmoko, myself and members of the Openmok
- community. For much of this time the primary Openmoko kernel maintaine
- was Andy Green
- Signed-off-by: Graeme Gregory <graeme@openmoko.com
- Signed-off-by: Andy Green <andy@openmoko.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix lm4857 contro
- Commit 4eaa9819dc08d7bfd1065ce530e31b18a119dcaf changed semantics o
- private_value member of kcontrol. This resulted in inability to contro
- amplifier and subsequently in very low output volume
- Tested-by: Johannes Schauer <josch@pyneo.org
- Signed-off-by: Paul Fertser <fercerpav@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - [ARM] S3C24XX: GPIO: Move gpio functions out of <mach/hardware.h
- Move all the gpio functions out of <mach/hardware.h> a
- this file is for defining the generic IO base addresse
- for the kernel IO calls
- Make a new header <mach/gpio-fns.h> to take this an
- include it via the chain from <linux/gpio.h> which i
- what most of these files should be using (and will b
- changed as soon as possible)
- Note, this does make minor changes to some drivers bu
- should not mess up any pending merges
- CC: Richard Purdie <rpurdie@rpsys.net
- Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- CC: David Brownell <dbrownell@users.sourceforge.net
- Signed-off-by: Ben Dooks <ben-linux@fluff.org
- - [ARM] S3C24XX: Remove hardware specific registers from DM
- call
- The S3C24XX DMA API channel configuration registers are being passe
- values comprised of register values which makes it hard to move th
- API to cover both the S3C24XX and S3C64XX
- These values can be calculated from knowing which device the channe
- is connected to, so remove them from the two calls s3c2410_dma_confi
- and s3c2410_dma_devconfig
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Ben Dooks <ben-linux@fluff.org
- - ASoC: Use platform device resource for S3C64xx IISv
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Staticise txctrl and rxctrl for S3C IISv
- They aren't used by anything external and aren't prototyped; if an
- users appear they can be exported again for them
- Also report what modes we have a problem with when we encounter invali
- mode configurations
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Display S3C IISv2 mode and MS errors by defaul
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Display the clock rate used as the basis for rate calculatio
- Aids debugging
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Allow use of resource from the platform device for S3C IISv
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix boot warnings from S3C IISv
- On startup we try to make sure that the port is quiesced but if th
- port is already stopped then this will generate a warning about th
- RX/TX mode configuration. Configure the mode before doing the teardow
- to suppress these warnings
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix data format configuration for S3C64xx IISv2 and add 24 bi
- The data format configuration for S3C64xx IISv2 is completely differen
- to that for S3C24xx. Instead of a single bit configuration in bit 0 o
- IISMOD we have format selection in bits 13 and 14 and bit clock rat
- selection in bits 1 and 2. While we're here add support for 24 bi
- samples in S3C64xx
- At some point it may be desirable to expose the bit clock rate selectio
- to users but given the limited configuration options that may not b
- required
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Make S3C64xx clock export function to return struct cl
- This makes the interface usable with the s3c-iis-v2 rate calculato
- and consistent with S3C2412
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Check for supported CPUs when building s3c-i2s-v
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix error message formatting in s3c64xx-i2s drive
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Use our registration function for S3C64x
- Make sure we get the DAI operations initialised
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: s3c-i2s-v2 diagnostic improvement
- Say what invalid values we're seeing when we see an invalid value an
- ensure that errors are displayed by default
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Enforce symmetric rates for S3C64xx I2S interfac
- There is only one LRCLK pin on each interface
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: S3C2412: Failing to get the I2S clock is an erro
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix S3C64xx IIS device registration and support both port
- The S3C64xx IIS code had a number of problems with device registration
- The hardware has two IIS ports of which the driver supported only on
- at once via a single exported DAI, attempting to identify the DAI t
- use based on the dev->id of the ASoC platform device. As well a
- limiting the driver to only supporting one IIS port at once this als
- meant that the ID of the soc-audio device (or in future the card device
- had to match the IIS ID
- Fix both problems by converting the driver to register the DAIs based o
- probing of platform devices registered by the arch/arm code, using thos
- platform devices to interact with the clock API
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Blackfin
- - ASoC: Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm da
- 1. fix "line over 80 characters" checkpatch warning
- 2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instea
- 3. fix typo
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: board driver to connect bf5xx with ad193
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: blackfin I2S(TDM mode) CPU DAI drive
- The I2S DAI driver for blackfin SPORT, but works in TDM mode
- I2S is not a special case of TDM with only left and right two slots fo
- SPORT interface. I2S coordinates with TDM in SPORT, but not a part o
- TDM. TDM require different hardware configuration with I2S, not onl
- different slot number. One is "Stereo Serial Operation" mode of SPORT
- the other one is "Multichannel Operation" mode. They are incompatibl
- at the same time
- Hardware and DMA description and data transfer flow are much differen
- for I2S and TDM. Merging them as a whole will be very ugly and difficul
- to maintain
- So we don't define a new DAI type, but give two DAI instances for standar
- I2S and TDM, both in I2S-family DAI type. The TDM instance still uses th
- I2S-family DAI type
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Blackfin I2S: fix resume handlin
- There is no need to manually start playback/capture ourselves as the PC
- driver will handle things for us
- Signed-off-by: Cliff Cai <cliff.cai@analog.com
- Signed-off-by: Mike Frysinger <vapier@gentoo.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Blackfin AC97: fix resume handlin
- There is no need to manually start playback/capture ourselves as the PC
- driver will handle things for us
- Signed-off-by: Cliff Cai <cliff.cai@analog.com
- Signed-off-by: Mike Frysinger <vapier@gentoo.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Blackfin: convert internal names from bf52x to bf5x
- These drivers aren't BF52x specific, so don't use bf52x in the names
- Signed-off-by: Barry Song <barry.song@analog.com
- Signed-off-by: Mike Frysinger <vapier@gentoo.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Blackfin: update the bf5xx_i2s_resume parameter
- Latest ASoC only passes snd_soc_dai to the resume function
- Signed-off-by: Barry Song <barry.song@analog.com
- Signed-off-by: Mike Frysinger <vapier@gentoo.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: Blackfin: keep better track of SPORT configuration stat
- Do not let the SPORT be reconfigured until there are no more activ
- streams. Then we can let the system reprogram the SPORT state
- Signed-off-by: Cliff Cai <cliff.cai@analog.com
- Signed-off-by: Mike Frysinger <vapier@gentoo.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: Blackfin: document how anomaly 05000250 is handle
- Signed-off-by: Sonic Zhang <sonic.zhang@analog.com
- Signed-off-by: Mike Frysinger <vapier@gentoo.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Blackfin: set the transfer size according the ac97_frame siz
- Signed-off-by: Cliff Cai <cliff.cai@analog.com
- Signed-off-by: Mike Frysinger <vapier@gentoo.org
- Signed-off-by: Bryan Wu <cooloney@kernel.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec AC97
- - ASoC: Use a shared define for AC97 CODEC data format
- The AC97 wire format is completely fixed so CODECs don't have any choic
- about the formats they accept but controllers accept a variety of dat
- formats and render them down onto the bus. Have a shared define so al
- the CODEC drivers will interoperate with any of our controller drivers
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec AD1836
- - Add more missing build stubs for ASo
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: Minor cleanups to AD1938 drive
- - Build in SND_SOC_ALL_CODECS
- - Remove null suspend/resume stuff
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: new ad1836 codec driver based on aso
- There has been an ad1836 driver in sound/blackfin based on traditional alsa
- The new driver is based on asoc. The architecture of ad1836 codec driver i
- very much like ad1938
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec AD1938
- - ASoC: delete -spi suffix in ad1938 and free private data while registers fai
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: add output/input widgets in ad1938 to make dac/adc dynamic PM wor
- According to the function dapm_dac_check_power() i
- sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without an
- output widget as sink. And according to dapm_adc_check_power(), ad
- power can't be on/off stand-alone without any input widget as source. S
- we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_AD
- to hope their power can be managed dynamically
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Update AD1938 for new TDM slot AP
- It's only actually paying attention to the slot count anyway
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix checkpatch issues and typos of ad1938 codec and bf5xx-tdm da
- 1. fix "line over 80 characters" checkpatch warning
- 2. ‘DMA_nnBIT_MASK’ is deprecated, use DMA_BIT_MASK instea
- 3. fix typo
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix checkpatch issues in AD193
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Kill direct accesses to driver_dat
- Replaced with dev_{get|set}_drvdata()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: new ad1938 codec driver based on aso
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec AD1980
- - ASoC: Use a shared define for AC97 CODEC data format
- The AC97 wire format is completely fixed so CODECs don't have any choic
- about the formats they accept but controllers accept a variety of dat
- formats and render them down onto the bus. Have a shared define so al
- the CODEC drivers will interoperate with any of our controller drivers
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec AK4535
- - ASoC: Remove unused AK4535 hardware read functionalit
- Nothing uses it and the existing hw_read operation needs to b
- refectored so it's easier to remove it rather than work with it
- Support can be re-added if the code requires volatile registers
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec AK4642
- - ASoC: Add ak4642/ak4643 codec suppor
- This is very simple driver for ALS
- It supprt headphone output and stereo input onl
- This patch is tested by ms7724s
- Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec CS4270
- - ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_devic
- Power management for the cs4270 codec is currently implemented as par
- of the i2c_driver struct. The disadvantage of doing it this way is tha
- the callbacks registered in the snd_soc_card struct are called _before
- the codec's callbacks
- That doesn't work, because the snd_soc_card callbacks will most likel
- switch down the codec's power domains or pull the reset GPIOs, an
- hence make the i2c communication bail out
- Fix this by binding the suspend and resume code to th
- snd_soc_codec_device driver model and let the I2C functions only cal
- the SoC core function for resume and suspend, which do nothing currentl
- but will do later
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Cc: Timur Tabi <timur@freescale.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: cs4270: add power management suppor
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Acked-by: Timur Tabi <timur@freescale.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: cs4270: introduce CS4270_I2C_INC
- Replace the magic 0x80 value with a suitable macro definition
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Acked-by: Timur Tabi <timur@freescale.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: cs4270: add Master Playback Switc
- This adds a new control named 'Master Playback Switch' for cs427
- codecs. It is implemented using the new SOC_DOUBLE_EXT macro to catc
- the put function and store the information about manually set mut
- controls from userspace. When a manual mute is set, we don't want th
- soc core to un-mute the outputs
- Renamed cs4270_mute() to cs4270_dai_mute() to avoid confusion
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Acked-by: Timur Tabi <timur@freescale.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: cs4270: fix Master Capture Switch polarit
- The control modifies the MUTE register, hence the polarity must b
- inverted
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Acked-By: Timur Tabi <timur@freescale.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec CX20442
- - ASoC: CX20442: simplify codec controller usag
- This patch is a workaround for the problem of several subsequent contro
- statements not being applied correctly to the codec controller (modem)
- In order to follow the hook switch state change from handset to handsfre
- whil
- in full duplex mode, two consecutive +VLS control commands were sent to th
- modem. The first one was M1 (microphone only), the seconds one was M1S1 (bot
- microphone and speaker). As there was no real modem handshaking procedur
- implemented, neither in the codec nor in the machine driver part of the lin
- discipline, the modem was having the second command missed
- Since a possibility to switch to microphone only mode (and speaker only mod
- as well) seams of no value, I have modified the code to issue single M1S
- command only for any of those cases
- Tested on my Amstrad Delta
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: CX20442: add some debuggin
- This patch adds debugging statement that can help in tracin
- how the driver is trying to control the codec device
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: CX20442: push down machine independent line discipline bit
- This corrected patch adds machine independent line discipline code, prevoiusl
- exsiting inside my Amstrad Delta ASoC machine dirver, to the Conexant CX2044
- codec driver. The code can be used as a standalone line discipline, or as
- set of codec specific functions called from machine's line disciplin
- callbacks. Anyway, the line discipline itself must be registered by a machin
- driver
- Applies on top of the followup to my initial driver version
- http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019757.htm
- Suggested by ASoC manintainer Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: CX20442: fix issues pointed out by subsystem maintaine
- The patch fixes some checkpatch identified issues and adds a comment abou
- line discipline interaction to my driver code, as requested by Mark on m
- inital submission (thank you Mark for applying my imperfect patch anyway)
- It also fixes MODULE_ALIAS mismatch as used in my machine driver
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add support for Conexant CX20442-11 voice modem code
- This patch adds support for Conexant CX20442-11 voice modem codec, suitabl
- for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Relate
- sound card driver will follow
- This codec is an optional part of the Conexant SmartV three chip modem design
- As such, documentation for its proprietary digital audio interface is no
- available. However, on Amstrad Delta board, thanks to Mark Underwood wh
- created an initial, omap-alsa based sound driver a few years ago[1], the code
- has been discovered to be accessible not only from the modem side, but als
- over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any soun
- card that can access the codec DAI directly. The DAI configuration parameter
- (sample rate and format, number of channels) has been selected out empiricall
- for best user experience
- The codec analogue interface consists of two pairs of analogue I/O pins
- speakerphone interface or telephone handset/headset interface. Furthermore, i
- seams to provide two operation modes for speakerphone I/O: standard an
- advanced, with automatic gain control and echo cancelation. Even if the code
- control interface is unknown and not available, all those interfaces and mode
- can be selected over the modem chip using V.253 commands. The driver is abl
- to issue necessary commands over a suitable hw_write function if provided by
- sound card driver. Otherwise, the codec can be controlled over the modem fro
- userspace while inactive
- Even if nothig is known about the codec internal power managemen
- capabilities, DAPM widgets has been used to model the codec audio map
- Automatically performed powering up/down of those virtual widgets results i
- corresponding V.253 commands being issued
- Some driver features/oddities may be board specific, but I have no way t
- verify that with any board other than Amstrad Delta
- [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.htm
- Created and tested against linux-2.6.31-rc3
- Applies and works with linux-omap-2.6 commi
- 7c5cb7862d32cb344be7831d466535d5255e35ac as well
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec DIT SPDI/F
- - ASoC: spdif: set module licence to GP
- Without MODULE_LICENCE("GPL"), when built as a module it will fai
- to load because it uses other GPL symbols from kernel
- Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: spdif codec: enable use by module
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Initialise dev for the dummy S/PDIF DA
- Also include the header to make sure the DAI is prototyped
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add dummy S/PDIF codec suppor
- McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed
- This patch provides stub codec that can be used in these configurations
- On DM646x EVM the McASP1 is connected to the S/PDIF out
- Signed-off-by: Steve Chen <schen@mvista.com
- Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com
- Signed-off-by: Naresh Medisetty <naresh@ti.com
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec MAX9877
- - ASoC: MAX9877: fix write operation for registe
- The MAX9877 needs an address of start register when we write values t
- registers through i2c_master_send(), but the code for this was missed i
- max9877_write_regs()
- If the value of control is 0 in the max9877_set_out_mode(), the value i
- not increased to 1, but actually the value to write to the registe
- should be 1
- And the register bits for out_mode and osc_mode should be cleared befor
- writing
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: MAX9877: separate callback function
- The callback function to control register was used by whole controls i
- MAX9877 driver, but this causes using many if statement for doubl
- register control or invert
- So, the callback function for double register control is separat
- differently, and the code for invert is added in the callback function
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: MAX9877: add MAX9877 amp drive
- The MAX9877 combines a high-efficiency Class D audio power amplifie
- with a stereo Class AB capacitor-less DirectDrive headphone amplifier
- The max9877_add_controls() is called to register the MAX9877 specifi
- controls on machine specific init() of the machine driver
- The datasheet for the MAX9877 can find at the following url
- http://datasheets.maxim-ic.com/en/ds/MAX9877.pd
- [Slight edit to sort the ALL_CODECS entries -- broonie.
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec Philips UDA134x
- - ASoC: UDA134X: Fix mistaken mute/unmute cod
- There is a mistake in current uda134x_mute function: mute_reg has bee
- changed in line 162 or line 164, so uda134x_write should writ
- "mute_reg" but not "mute_reg & ~(1<<2)" t
- UDA134X_DATA010
- Signed-off-by: Shine Liu <shinel@foxmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec Philips UDA1380
- - ASoC: UDA1380: refactor device registratio
- This patch mostly follows commit 5998102b9095fdb7c67755812038612afea315c
- "ASoC: Refactor WM8731 device registration" to make UDA1380 use standar
- device instantiation. Similarly, the I2C device registration temporaril
- moves into the magician machine driver before it will find its fina
- resting place in the board file
- At the same time, platform specific configuration is moved to platform dat
- and common power/reset GPIO handling moves into the codec driver
- Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec SSM2602
- - ASoC: Revert duplicated code in SSM2602 drive
- The Blackfin submission was done as a patch against a different tre
- and contained a duplicate hunk which will cause us to loose track of th
- substream pointers when shutting down. Remove one of the duplicate
- hunks
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: SSM2602: assign last substream to the master when shutting dow
- Fixes crash when shutting down
- Signed-off-by: Cliff Cai <cliff.cai@analog.com
- Signed-off-by: Mike Frysinger <vapier@gentoo.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: SSM2602: remove unsupported sample rate
- Signed-off-by: Cliff Cai <cliff.cai@analog.com
- Signed-off-by: Mike Frysinger <vapier@gentoo.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec STAC9766
- - ASoC: Keep index within stac9766_reg[
- Keep index within stac9766_reg[
- Signed-off-by: Roel Kluin <roel.kluin@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix minor issues in STAC9766 drive
- Fairly minor issues
- - Don't register the DAIs, it's not required for AC97 devices
- - Make unexported functions static
- - Wrap some excessively long lines
- - Undo tab/space breakage
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Codec for STAC9766 used on the Efik
- Datasheet: http://www.idt.com/products/getDoc.cfm?docID=1313400
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec TLV320AIC23
- - ASoC: codec tlv320aic23 fix bogus divide by 0 messag
- Some code analyzer software mistakenly give
- divide by 0 error messages for these lines
- This patch will end its confusion
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: correct print specifiers for unsigned
- Unsigned variables should use `%u' rather than `%d'
- Signed-off-by: Roel Kluin <roel.kluin@gmail.com
- Signed-off-by: Andrew Morton <akpm@linux-foundation.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: tlv320aic23: add DSP_A format suppor
- Add DSP_A interface format support by setting the LRP bit i
- DSP mode
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec TLV320AIC3X
- - ASoC: Make platform data optional for TLV320AIC3
- Now that we don't need the I2C address for the device the platform dat
- is redundant so allow it to be omitted
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Tested-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: tlv320aic3x: Change to use device mode
- The tlv320aic3x driver managed its own i2c device, instead of an extan
- one created by the board support code. Change the code to make it so tha
- the driver binds to an extant (in this case i2c) device
- Add explict tlv320aic33 as well as tlv320aic3x to the supported devic
- table and remove the old driver bindings from the users of this code
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove use of hw_read from TLV320AIC3x drive
- The TLV320AIC3x driver is currently the only user of the CODEC hw_rea
- operation and is jumping through some hoops in order to do so. In orde
- to support future refactoring to make the hw_read operation more usabl
- unwrap the usage in this driver to avoid its use
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: tlv320aic3x: Enable PLL when not bypasse
- PLL was not being enabled when it was not bypassed. This patc
- enables the PLL when it is used. Additionally, it disables the PL
- when it is bypassed
- Without this patch, the audio on TI DM646x EVM and DM355 EV
- does not work properly. The bit clocks and the frame sync signal
- from the codec are not correct and hence the playback/record are faste
- than usual for most sample rates. The reason for this was that the PL
- was not enabled when it was not bypassed
- Tested on DM6467 EVM, playback tested on DM355 EVM
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
SoC Codec TWL4030
- - ASoC: TWL4030: Fix for capture mixer string
- Change the strings related to capture in order to b
- interpreted correctly by alsamixer and possible othe
- UI based mixer applications
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Introduce PGAs for output
- Dynamically control and control only the needed output amplifie
- muting/un-muting
- The original code was muting and un-muting the following outpu
- amplifiers: Earpiece PreDrivL/R, CarkitL/R at the same tim
- regardless which pin is actually in use at the given moment
- Move these as separate PGA so only the needed amplifier will be touched
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add tristate callbacks for HiFi and Voic
- Add "set_tristate" callbacks for HiFi and Voice DAIs
- Machine drivers can enable and disable tristate for eac
- DAI with "snd_soc_dai_set_tristate" function
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add EXTMUTE to reduce pop-noise effec
- According to TRM, an external FET controlled by a 1.8V output signa
- can be used to reduce the pop-noise heard when the audio amplifier i
- switched on. It is suggested that GPIO6 of TWL4030 be used, but an
- other gpio can be used instead. This is indicated in machine drive
- with the following twl4030_setup_data members
- -hs_extmute. Set to 1 if board has support for EXTMUTE
- -set_hs_extmute. Set to a callback funcion to control an external gpi
- line. Set to NULL if MUTE[GPIO6] pin is used
- Codec driver takes care of enabling and disabling this output durin
- the headset pop attenuation sequence
- Also add a delay to let VMID settle in ramp up sequence
- Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove word "Switch" from Handsfree switch nam
- SoC dapm adds the suffix "Switch" to SND_SOC_DAPM_SWITCH controls
- removing word "Switch" from HandsfreeL/HandsfreeR widget nam
- for avoiding to duplicate it
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Correct bypass event for voice sideton
- Event for voice sidetone was being interpreted as a
- analog HiFi bypass event because VSTPGA register offse
- is less than ARXR2_APGA_CTL offset. Reordering th
- register checks allows to handle voice digital bypas
- event properly
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add AVADC Clock Priorit
- AVDAC clk priority allows to determine the path ADC mus
- be connected when the codec is in option2 and both HiF
- and Voice paths are enabled
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Fix voice interface clock master
- Voice interface of twl4030 codec supports: CBM_CFM an
- CBS_CFS. It doesn't support CBS_CFM
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-By: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Staticise put_twl4030_opmode_enum_double(
- It's an operation for a control and doesn't need to be exported
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix shadowed variables in twl403
- No need to define second copies of mode and format, they're declare
- with exactly the same type at the head of the function and there's n
- conflict in their use
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix build error in twl4030.
- Fix the (likely cut-n-paste) error by commi
- 16a30fbb0d3aa4ee829a2dd3d0e314e2b5ae96a9, which causes the error below
- sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache'
- sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Check the interface format for 4 channel mod
- In addition to the operating mode check, also check th
- codec's interface format in case of four channel mode
- If the codec is not in TDM (DSP_A) mode, return with error
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Use reg_cache in twl4030_init_chi
- Use the codec->reg_cache instead of the array directl
- in twl4030_init_chip for setting the default values
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: HandsfreeL/R mute DAPM switc
- Add DAPM switch for HeadsetL/R mute. Since all bits are are neede
- for the HFL/R pop removal to work the switch is using the SW_SHADO
- no HW register for the HandsfreeL/R mute
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add shadow registe
- Shadow, non HW register for dealing with the HandsfreeL/
- muting
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Handsfree pop removal redesig
- Move the HandsfreeL/R (IHFL/R) pop removal code from the DAPM_MUX_
- to a more appropriate DAPM_PGA_E widget
- Also fix the power-up sequence to match with the TRM
- The power-down sequence is not described in the TRM, so do i
- in a way, which seams like the correct sequence
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Differentiate the playback stream
- Give unique stream names for the two playback streams s
- DAPM can figure out which codec_dai is in use
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add support for platform dependent configuratio
- twl4030_setup_data structure can be passed from platform drivers t
- the codec via the snd_soc_device->codec_data pointer
- Currently the setup data has support for the Headset pop-remova
- related configuration, which differs from board to board
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Move the Headset pop-attenuation code to PGA even
- This patch adds SND_SOC_DAPM_PGA_E to the headset path, which handle
- the headset ramp up and down sequences needed for the pop nois
- removal
- With this patch the order of the internal components in the twl403
- codec is turned on and off in a correct order
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com
- Tested-by: Jarkko Nikula <jhnikula@gmail.com
- Tested-by: Misael Lopez Cruz <x0052729@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Change DAPM routings and controls for DACs and PGA
- Restructuring the twl4030 codec's DAPM routing to be able to handle the powe
- sequences correctly
- The twl4030 codec internal implementation have this order
- DAC -> Analog PGA -> Mixer/Mu
- While the ASoC framework expects the following order
- DAC -> Mixer -> Analog PG
- This patch moves the Analog PGA handling from SND_SOC_DAPM_PGA to _MIXER an
- adds two levels of mixer to handle the digital and analog loopbac
- functionality
- Now the analog loopback does not powers on any of the DACs
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Tested-by: Anuj Aggarwal <anuj.aggarwal@ti.com
- Tested-by: Jarkko Nikula <jhnikula@gmail.com
- Tested-by: Misael Lopez Cruz <x0052729@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add control for selecting codec operation mod
- Add a control for selecting the codec operation mode. TWL4030 code
- has two modes
- - Option 1. Audio only (4 audio DACs
- - Option 2. Voice/Audio (2 audio DACs and voice ADC/DAC
- Control is restricted when a stream is ongoing, since codec'
- operation mode cannot be changed on-the-fly
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Fix Analog capture path for AUX
- AUXR is selected by bit 2 and not by bit 1 in the ANAMICR register
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Enable/disable voice digital filter
- Enable TWL4030 VTXL/VTXR and VRX digital filters for uplin
- and downlink paths, respectively
- This patch also corrects voice 8/16kHz mode selection bi
- (SEL_16K) of CODEC_MODE register
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: change DAPM for analog microphone selectio
- The inputs of the twl4030 codec can be mixed, so we will use the mixe
- DAPM for the analog microphone registers(0x05, 0x06), but if we enabl
- more than one input at the same time, the input impedance of the inpu
- amplifier will be reduced
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Fix typo in twl4030_codec_mute functio
- Copy-paste error: TWL4030_PRECKL_GAIN >> TWL4030_PRECKR_GAI
- It has not caused problems, sinc
- TWL4030_PRECKL_GAIN == TWL4030_PRECKR_GAIN == 0x3
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add VIBRA outpu
- This patch adds support for the VIBRA output on TWL4030 codec
- The VIBRA output can be driven with audio data or wit
- local vibrator driver
- Add the needed DAPM elements and routes for the VIBRA output an
- controls for the VIBRA driver configuration
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add voice digital loopback: sideton
- This patch add voice digital loopback (sidetone) to the twl403
- driver. It mixes voice uplink attenuated (by sidetone gain) wit
- voice downlink when the codec is working in option2 (voice/audi
- mode)
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add VDL analog bypas
- This patch adds voice downlink analog bypass switch. It follow
- the same approach as in other analog bypass switches
- DAC switch is moved from 'DAC Voice' to 'Analog Voice Playback Mixer'
- that will also allow voice DAC to be powered in digital voic
- loopback (sidetone)
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add 4 channel TDM suppor
- Support for 4 channel TDM (SND_SOC_DAIFMT_DSP_A) for twl403
- codec
- The channel allocations are
- Playback
- TDM i2s TWL R
- Channel 1 Left SDRL
- Channel 3 Right SDRR
- Channel 2 -- SDRL
- Channel 4 -- SDRR
- Capture
- TDM i2s TWL T
- Channel 1 Left TXL
- Channel 3 Right TXR
- Channel 2 -- TXL
- Channel 4 -- TXR
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add VDL path suppor
- Add DAPMs for VDL(Voice Down Link) path. To support VDL path, we hav
- to change DAPMs of outputs(Earpiece, PreDrive Left/Right, Headse
- Left/Right, Carkit Left/Right) from mux to mixer
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add support Voice DA
- Add Voice DAI to support the PCM voice interface of the twl4030 codec
- The PCM voice interface can be used with 8-kHz(voice narrowband) o
- 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mon
- TX or stereo TX
- The PCM voice interface has two mode
- - PCM mode1 : This uses the normal FS polarity and the rising edge o
- the clock signal
- - PCM mode2 : This uses the FS polarity inverted and the falling edg
- of the clock signal
- If the system master clock is not 26MHz or the twl4030 codec mode is no
- option2, the voice PCM interface is not available
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Fix for the constraint handlin
- The original implementation of the constraints were good against san
- applications
- If the opening sequence is
- stream1_open, stream1_hw_params, stream2_open, stream2_hw_params -> th
- constraints are set correctly for stream2
- But if the sequence is
- stream1_open, stream2_open, stream2_hw_params, stream1_hw_params -> than stream
- would receive constraint rate = 0, sample_bits = 0, since the stream1 has no
- yet called hw_params..
- The command to trigger this event
- gst-launch-0.10 alsasrc device=hw:0 ! alsasink device=hw:0 sync=fals
- This patch does some 'black magic' in order to always set the correc
- constraints and sets it only when it is needed for the other stream
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Fix gain control for earpiece amplifie
- The gain control for earpiece amplifier uses 0dB ~ 12dB according to th
- TRM, but the present code is implemented to -6dB ~ 6dB
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
SoC Codec WM8350
- - ASoC: Don't reconfigure WM8350 FLL if not neede
- If the requested FLL configuration is the one we're currently runnin
- in it's at best pointless to reconfigure the FLL
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add suspend and resume callbacks to Wolfson CODEC driver
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Automatically manage WM8350 sloping stopband filte
- For best performance the DAC sloping stopband filter should be enable
- below 24kHz and not enabled above that so remove the user visibl
- control for this and do it autonomously in the driver
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Include WM8350 register definitions in CODEC heade
- It's expected behaviour for the CODEC header to provide them but th
- WM8350 doesn't due to having all the registers together under drivers/mfd
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix logic in WM8350 master clocking chec
- We need to check only if the WM8350 is master and only when startin
- the stream so if either is not true then we can skip the check
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
SoC Codec WM8400
- - ASoC: Bodge around GCC 4.4.0 flow analysis bug in GCC 4.4.
- GCC 4.4.0 doesn't appear to be able to spot that we don't apply any FL
- configuration if the output frequency is zero
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add suspend and resume callbacks to Wolfson CODEC driver
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: remove driver_data direct access of struct devic
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8510
- - ASoC: Factor out 7 bit register 9 bit data SPI writ
- This converts all the Wolfson drivers using this format (the only device
- that do) except WM8753 to use it
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add I/O control bus information to factored out cache setu
- While writes tend to be able to use a fairly bus independant format t
- do the writes reads are all bus specific. To allow us to factor ou
- this code include the bus type as a parameter when setting up th
- cache
- Initially just use this to factor out hw_write_t for I2C
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Begin to factor out register cache I/O function
- A lot of CODECs share the same register data formats and therefor
- replicate the code to manage access to and caching of the registe
- map. In order to reduce code duplication centralised versions o
- this code will be introduced with drivers able to configure the us
- of the common code by calling the new snd_soc_codec_set_cache_io(
- API call during startup
- As an initial user the 7 bit address/9 bit data format used by man
- Wolfson devices is supported for write only CODECs and the driver
- with straightforward register cache implementations are converted t
- use it
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: WM8510 has a single frame clock so needs symmetric rate
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8523
- - ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODEC
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8523 CODEC drive
- The WM8523 is a high performance stereo DAC with integral charg
- pump providing 2Vrms line driver outputs using a single 3.3V powe
- supply rail
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8580
- - ASoC: Add I/O control bus information to factored out cache setu
- While writes tend to be able to use a fairly bus independant format t
- do the writes reads are all bus specific. To allow us to factor ou
- this code include the bus type as a parameter when setting up th
- cache
- Initially just use this to factor out hw_write_t for I2C
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out WM8580 register cache cod
- Note the slightly tricky cache usage in the volume update function du
- to the requirement for a separate write for the VU bit
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Regulator support for WM858
- Add basic support for integration with the regulator API to WM8580
- Since the core cannot yet disable biases when the CODEC is idle w
- simply request and enable the regulators for the entire time th
- driver is active
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add suspend and resume callbacks to Wolfson CODEC driver
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8728
- - ASoC: Factor out 7 bit register 9 bit data SPI writ
- This converts all the Wolfson drivers using this format (the only device
- that do) except WM8753 to use it
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add I/O control bus information to factored out cache setu
- While writes tend to be able to use a fairly bus independant format t
- do the writes reads are all bus specific. To allow us to factor ou
- this code include the bus type as a parameter when setting up th
- cache
- Initially just use this to factor out hw_write_t for I2C
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Begin to factor out register cache I/O function
- A lot of CODECs share the same register data formats and therefor
- replicate the code to manage access to and caching of the registe
- map. In order to reduce code duplication centralised versions o
- this code will be introduced with drivers able to configure the us
- of the common code by calling the new snd_soc_codec_set_cache_io(
- API call during startup
- As an initial user the 7 bit address/9 bit data format used by man
- Wolfson devices is supported for write only CODECs and the driver
- with straightforward register cache implementations are converted t
- use it
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8731
- - ASoC: Drop unneeded declaration of removed wm8731 SPI write functio
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out 7 bit register 9 bit data SPI writ
- This converts all the Wolfson drivers using this format (the only device
- that do) except WM8753 to use it
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Limit WM8731 to symmetric rate
- While the hardware is capable of some limited asynmmetric modes th
- driver does not currently support those modes so tell application
- that only symmetric rates are available
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Correct WM8731 Mic Capture Switch control nam
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add TLV information for WM873
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix leaks in WM8731 probe error handlin
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add suspend and resume callbacks to Wolfson CODEC driver
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: remove driver_data direct access of struct devic
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8750
- - ASoC: Factor out 7 bit register 9 bit data SPI writ
- This converts all the Wolfson drivers using this format (the only device
- that do) except WM8753 to use it
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8753
- - ASoC: Fix wm8753 register cache size and initializatio
- Register cache space was not being allocated for the final register
- causing bugs when it was used. Allocate space for it
- Also ensure that the final register is displayed in sysfs
- [Commit message rewritten to document actual issue. -- broonie
- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix register cache initialisation for WM875
- The wrong register cache variable was being used to provide the size fo
- the memcpy(), resulting in a copy of only a void * of data
- Reported-by: Lars-Peter Clausen <lars@metafoo.de
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Cc: stable@kernel.or
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: remove driver_data direct access of struct devic
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8776
- - ASoC: Convert WM8776 to use factored out register cache cod
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8776 CODEC drive
- The WM8776 is a high performance, stereo audio CODEC with five channe
- input selector. The WM8776 is ideal for surround sound processin
- applications for home hi-fi, DVD-RW and other audio visual equipment
- This driver implements support for most WM8776 features - currently th
- ADC automatic level control/limiter functionality is omitted
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8900
- - ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODEC
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Automatically manage WM8900 sloping stopband filte
- For best performance the DAC sloping stopband filter should b
- enabled below 24kHz and not enabled above that so remove th
- user visible control for this and do it autonomously in th
- driver
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8903
- - ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODEC
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Automatically control WM8903 sloping stopband filte
- For best performance the DAC sloping stopband filter should b
- enabled below 24kHz and not enabled above that so remove th
- user visible control for this and do it autonomously in th
- driver
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove odd bit clock ratios for WM890
- These are not supported since performance can not be guarantee
- when they are in use
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Cc: stable@kernel.or
- - ASoC: Implement WM8903 digital sidetone suppor
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove redundant rate constraint for WM890
- This is now handled by symmetric_rates
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Actively manage the DC servo for WM890
- Save a little extra power by enabling the DC servo offset correctio
- for the output channels only when the relevant channels are enabled
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Optimise configuration of WM8903 DC serv
- Modify the default startup sequence in the chip to set the DC serv
- dither level for optimal performance
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Support CLK_DSP in WM890
- CLK_DSP provides a master clock for the DAC and ADC related functionalit
- on the device
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Use DAPM supply widget for WM8903 charge pum
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Request shared rates for WM890
- It has a shared LRCLK
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8940
- - ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODEC
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add missing __devexit in wm8940.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Staticise TLV values in WM894
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: ASoC WM8940 Drive
- Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8960
- - ASoC: Fix WM8960 leaks on probe failur
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8960 CODEC drive
- The WM8960 is a low power, high quality stereo codec designed fo
- portable digital audio applications
- Stereo class D speaker drivers provide 1W per channel into 8W loads
- Guaranteed low leakage, excellent PSRR and pop/click suppressio
- mechanisms enable direct battery connection for the speaker supply
- The device also integrates a complete microphone interface and a stere
- headphone driver. External component requirements are drasticall
- reduced as no separate microphone, speaker or headphone amplifiers ar
- required. Advanced on-chip digital signal processing performs automati
- level control for the microphone or line input
- Stereo 24-bit sigma-delta ADCs and DACs are used with low powe
- over-sampling digital interpolation and decimation filters and
- flexible digital audio interface
- The master clock can be input directly or generated internally by a
- onboard PLL, supporting most commonly-used clocking schemes
- This driver was originally written by Liam Girdwood, with substantia
- subsequent additions and updates for feature completeness and changes i
- the ASoC framework from me
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8961
- - ASoC: Fix WM8961 suspend function typ
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add core suspend and resume callbacks to WM896
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8961 drive
- The WM8961 is a low power, high quality stereo CODEC designed fo
- portable digital applications with headphone and stereo class D speake
- drivers
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8974
- - Add more missing build stubs for ASo
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: Factor out cache I/O from WM897
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Correct a bug with "ADC Inversion Switch" in wm8974 codec
- This corrects a bug with ADC Inversion Switch in wm8974 codec
- Signed-off-by: Javier Martin <javier.martin@vista-silicon.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: WM8974 DAPM cleanup
- Also implement AUX mode control
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: WM8974 cosmetic cleanup
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Use symmetric rates for WM897
- The chip has a single LRCLK
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8974 TLV informatio
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Refresh WM8974 PLL configuratio
- Move away from a fixed table to runtime calculation
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Declare 2 channels for WM897
- The device is a mono device but it can read two channel data an
- many I2S controllers only understand 2 channels
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Refresh WM8974 bias configuratio
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove unreferenced wm8974_add_controls(
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Update WM8974 to use standard I2C device probe method
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: WM8974 checkpatch cleanup
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8974 CODEC drive
- The WM8974 is a low power, high quality mono CODEC designed for portabl
- applications such as digital still cameras or digital voice recorders
- This driver was originally written by Graeme Gregory and Liam Girdwoo
- and has since been maintained by myself with some updates contributed b
- Brett Saunders and Javier Martin
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8988
- - Sound: remove direct access of driver_dat
- This is the last in-kernel direct usage of driver_data, replace it wit
- the proper dev_get/set_drvdata() calls
- Cc: Takashi Iwai <tiwai@suse.de
- Cc: Jaroslav Kysela <perex@perex.cz
- Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Cc: Liam Girdwood <lrg@slimlogic.co.uk
- Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de
- - ASoC: Fix leaks in WM8988 registration error handlin
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8988 CODEC drive
- The WM8988 is a low power, high quality stereo CODEC designed fo
- portable digital audio applications
- The device integrates complete interfaces to 2 stereo headphone or lin
- out ports. External component requirements are drastically reduced as n
- separate headphone amplifiers are required. Advanced on-chip digita
- signal processing performs graphic equaliser, 3-D sound enhancement an
- automatic level control for the microphone or line input
- The WM8988 can operate as a master or a slave, with various master cloc
- frequencies including 12 or 24MHz for USB devices, or standard 256f
- rates like 12.288MHz and 24.576MHz. Different audio sample rates such a
- 96kHz, 48kHz, 44.1kHz are generated directly from the master cloc
- without the need for an external PLL
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM8990
- - ASoC: Fix errors in WM899
- The mis-typing exist in dapm controller definitions and dapm route definitions
- so happen mis-matched error when snd_soc_dapm_add_routes()
- Cc: stable@kernel.or
- Signed-off-by: Jinyoung Park <parkjy@mtekvision.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.co
SoC Codec WM8993/4
- - Add more missing build stubs for ASo
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: Remove unneeded inclusion of linux/regulator/consumer.
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove duplicate ADC/DAC widgets from wm_hubs.
- These need to be in the CODEC since the DAIs supported by the CODEC
- aren't static
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: WM8993 digital mixing suppor
- The WM8993 provides digital sidetone paths and also allows eac
- channel on the audio interface to be routed separtately to th
- DACs and ADCs
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Implement TDM configuration for WM899
- Note that the number of slots used internally is specified in term
- of stereo slots while the external API works with mono slots
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix WM8993 MCLK configuration for high frequency MCLK
- When used without the PLL we were accidentally clearing the MCLK/
- divider, resulting in a double rate SYSCLK when the divider shoul
- have been used
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out shared code from WM899
- The WM8993 analogue control is shared with other devices in the sam
- product line. Since this is a very substantial proportion of th
- driver move the definitions of these controls into a new wm_hubs modul
- which allows them to be shared between the two
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix FLL reference clock division setup in WM899
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix sample rate lookup in WM899
- We need to use the best value we picked, not the last value w
- looked at
- Reported-by: Stephen Rothwell <sfr@canb.auug.org.au
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8993 CODEC drive
- The WM8993 is a highly integrated ultra-low power hi-fi CODEC designe
- for portable devices such as multimedia phones
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM9081
- - ASoC: Update WM9081 for tdm_slot() API chang
- Store the TDM slot width then if it's set use that rather than th
- sample size to calculate BCLK. Leave imposing constraints to th
- core (which should do this but doesn't yet) or machine driver
- Also allow 0 TDM slots to be configure (for use when disabling TDM)
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: change set_tdm_slot api to allow slot_width override
- Extend set_tdm_slot to allow the user to arbitrarily set the frame widt
- and active TX/RX slots
- Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.
- still doesn't handle the slot_width override
- While being there, correct an incorrect use of SlotsPerFrm(7) use i
- bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) )
- (this series is meant for Mark's for-2.6.32 branch
- Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Error out if we can't determine a suitable WM9081 syscl
- Due to the flexibility of the WM9081 FLL this should never happe
- in a real system
- Reported-by: Jaswinder Singh Rajput <jaswinder@kernel.org
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix WM9081 PowerPC compiler issue
- Ensure that we always set a new sysclk when using the FLL in master mod
- and pick out the correct value for the sample rate in hw_params()
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier drive
- The WM9081 is designed to provide high power output at low distortio
- levels in space-constrained portable applications
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM9705
- - ASoC: free socdev if init_card() fails in wm9705_soc_probe(
- Free socdev if snd_soc_init_card() fails
- Signed-off-by: Roel Kluin <roel.kluin@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Use a shared define for AC97 CODEC data format
- The AC97 wire format is completely fixed so CODECs don't have any choic
- about the formats they accept but controllers accept a variety of dat
- formats and render them down onto the bus. Have a shared define so al
- the CODEC drivers will interoperate with any of our controller drivers
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM9712
- - ASoC: Support AC97 link off by default on WM971
- The WM9712 can be configured by resistor strapping GPIO4 to behave lik
- the WM9713 and default to leaving the AC97 link disabled after col
- reset until a warm reset occurs. In this configuration we need to issu
- a warm reset after cold to bring the link up so do so. The warm rese
- will be harmless on systems that don't need it
- [Changelog rewritten to document the reasoning. -- broonie
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Codec WM9713
- - ASoC: Move the WM9713 voice DAC powerdown to a DAPM even
- This ensures that we sync with the DAPM powerdown sequencing properl
- and don't need to bounce the power on the voice DAC so often
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: WM9713 requires symmetric rates on the voice DA
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC DaVinci
- - ASoC: tlv320aic3x: fixup board device change
- Fixup the device changes by modifying the files that we just removed th
- explicit device creation from with i2c_register_board_info() until thi
- can be moved into the relevant board files
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: tlv320aic3x: Change to use device mode
- The tlv320aic3x driver managed its own i2c device, instead of an extan
- one created by the board support code. Change the code to make it so tha
- the driver binds to an extant (in this case i2c) device
- Add explict tlv320aic33 as well as tlv320aic3x to the supported devic
- table and remove the old driver bindings from the users of this code
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EV
- There is one instance of McASP on DA850/OMAP-L138 SoC. This i
- connected to TLV320AIC3106 codec for audio playback and capture
- This patch adds audio support on this platform. Some of th
- structure prefix names which are common for DA830/OMAP-L137 EVM an
- DA850/OMAP-L138 EVM have been renamed to da8xx from da830
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: Add a DAI format to McASP drive
- The patch adds a DAI format: Codec bit clock master and frame sync slave
- to the driver
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: McASP driver enhacement
- On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIF
- support. This FIFO provides additional data buffering. It also provide
- tolerance to variation in host/DMA controller response times
- The read and write FIFO sizes are 256 bytes each. If FIFO is enabled
- the DMA events from McASP are sent to the FIFO which in turn sends DMA request
- to the host CPU according to the thresholds programmed
- More details of the FIFO operation can be found a
- http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber
- sprufm1&fileType=pd
- This patch adds support for FIFO configuration. The platform data has
- version field which differentiates the McASP on different SoCs
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: Support Audio on DA830 EV
- Add support for audio on DA830 EVM- here McASP1 is interfaced t
- TLV320AIC3106 codec
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: pcm, constrain buffer size to multiple of perio
- The dma setup code assumes that the buffer size is a multipl
- of the period size
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: i2s: don't bounce through rtd to get da
- dai is a parameter to the functions, so use it instead o
- looking it up
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: davinci: don't use clock name
- clock name strings are no longer passed on platform_data. Instead
- we rely entirely on struct device and clkdev to find the right clock
- Signed-off-by: Kevin Hilman <khilman@deeprootsystems.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Introduce platform driver model for dm644x, dm35
- Introduce the platform driver model to get platform data for dm355 and dm644x
- Register platform driver and acquire the resources in the probe function Sinc
- the platform specific code had been moved from machine driver to dm<soc>.
- Signed-off-by: Naresh Medisetty <naresh@ti.com
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci I2S needs mach/asp.
- Reported-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: pcm, don't play 1st sound period twic
- Update the dma link with correct data as soon a
- the master channel has copied it. Otherwise, th
- 1st period will play twice
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add machine driver support for DM646
- This patch does the following
- (1) Add support for the DM646x machin
- (2) Modifications required to introduce the platform driver model to ge
- platform data for all the machines including dm355 and dm644x
- Signed-off-by: Steve Chen <schen@mvista.com
- Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com
- Signed-off-by: Naresh Medisetty <naresh@ti.com
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add mcasp support for DM646
- Adds driver support for the two instances of McASP on TI's DM646x
- The multichannel audio serial port (McASP) functions as a general-purpose audi
- serial port optimized for the needs of multichannel audio application
- (http://www.ti.com/litv/pdf/spruer1b)
- There are two instances of McASP on DM646x. The McASP0 module includes up to
- serializers that can be individually enabled to either transmit or receiv
- in different modes. The McASP1 module is limited with only 1 pinned-ou
- serializer that can be enabled to only transmit in DIT mode (neither receivin
- in any mode nor transmitting in either Burst or TDM mode is supported)
- McASP0 consists of transmit and receive sections that may operat
- synchronized, or completely independently with separate master clocks, bi
- clocks, and frame syncs, and using different transmit modes with differen
- bit-stream formats
- Signed-off-by: Steve Chen <schen@mvista.com
- Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com
- Signed-off-by: Naresh Medisetty <naresh@ti.com
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: i2s, add davinci_i2s_prepare and shutdow
- If the codec is master then prepare should cal
- mcbsp_start, not trigger
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: i2s, fix mcbsp_word_length updat
- Code previously just "ors" in this field without clearin
- first. Fix, by never reading this register
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: i2s, minor cleanup of davinci_i2s_startu
- Save a few lines of code
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: i2s, only start sample generator if neede
- Only start sample generator if needed, and mor
- cleanup on davinci_mcbsp_start
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: i2s cleanu
- Move variable declaration closer to use
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoc: DaVinci: i2s, minor cleanu
- Add davinci_mcbsp_dev as argument to davinci_mcbsp_star
- and davinci_mcbsp_stop
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: i2s toggle clock to complete rese
- Add toggle_clock function to complete i2s reset earlier
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci: i2s, remove MOD_REG_BIT macr
- No functional changes. Rename variable w to somethin
- more meaningful. Remove code obfuscating macro MOD_REG_BIT
- Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: DaVinci EVM board support buildfixe
- This is a build fix, resyncing the DaVinci EVM ASoC board cod
- with the version in the DaVinci tree. That resync include
- support for the DM355 EVM, although that board isn't yet i
- mainline
- (NOTE: also includes a bugfix to the platform_add_resource
- call, recently sent by Chaithrika U S <chaithrika@ti.com> bu
- not yet merged into the DaVinci tree.
- Signed-off-by: David Brownell <dbrownell@users.sourceforge.net
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: DaVinci I2S update
- This resyncs the DaVinci I2S code with the version in the DaVinc
- tree. The behavioral change uses updated clock interfaces whic
- recently merged to mainline. Two other changes include adding
- comment on the ASP/McBSP/McASP confusion, and dropping pdev->id i
- order to support more boards than just the DM644x EVM
- Signed-off-by: David Brownell <dbrownell@users.sourceforge.net
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: davinci-pcm buildfixe
- This is a buildfix for the DaVinci PCM code, resyncing it wit
- the version in the DaVinci tree. The notable change is usin
- current EDMA interfaces, which recently merged to mainline
- (The older interfaces never made it into mainline.
- NOTE: open issue, the DMA should be to/from SRAM; see chi
- errata for more info. The artifacts are extremely easy t
- hear on DM355 hardware (not yet supported in mainline), bu
- don't seem as audible on DM6446 hardwaare (which does hav
- mainline support)
- Signed-off-by: David Brownell <dbrownell@users.sourceforge.net
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
SoC Dynamic Audio Power Management
- - ASoC: add missing inclusion of debugfs.
- To fix compile errors
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add DAPM widget power decision debugfs file
- Currently when built with DEBUG DAPM will dump information abou
- the power state decisions it is taking for each widget to dmesg
- This isn't an ideal way of getting the information - it require
- a kernel build to turn it on and off and for large hub CODECs th
- volume of information is so large as to be illegible. When th
- output goes to the console it can also cause a noticable impac
- on performance simply to print it out
- Improve the situation by adding a dapm directory to our debugf
- tree containing a file per widget with the same information i
- it. This still requires a decision to build with debugfs suppor
- but is easier to navigate and much less intrusive
- In addition to the previously displayed information active stream
- are also shown in these files
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Provide default set_bias_level() implementatio
- If the CODEC does not provide a set_bias_level() then update th
- bias_level variable for it since other parts of the system expec
- that to be maintained
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add input and output AIF widget
- Currently DAPM interfaces with the audio streams to and from th
- processor at the DAC and ADC widgets. As the digital capabilitie
- of parts increases this is becoming a less and less able to mee
- the needs of parts
- To meet the needs of these devices create new widgets interfacin
- with the TDM bus but not integrated into any other functionality
- Audio can then be routed to and from these widgets using existin
- routing widgets
- A slot number is provided in the definition but this is currentl
- not used yet. This is intended to support devices which can us
- more than one TDM slot on a single interface
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Power speakers and headphones simultaneousl
- Speaker and headphone outputs do not need to be handled separatel
- since they can't be part of the same path
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix handling of bias levels for non-DAPM codec
- If the system doesn't have any DAPM widgets then we can't use thei
- state to check if the bias level for the codec should be up
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: fix checking for external widgets bu
- In SOC DAPM layer of SOUND subsystem, when add signal route (in th
- function snd_soc_dapm_add_route() ), the original code has wrong logi
- when dapm layer check each widget whether an external one
- Signed-off-by: Rongrong Cao <rrcao@ambarella.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add pop delay debug at end of DAPM sequencin
- Provide an interval after the end of DAPM sequencing so that w
- can distinguish between a pop in the final step of the sequenc
- and a pop generated from some other source outside DAPM
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix widget powerdown on shutdow
- We need to set the widget power state we want to implement
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add a shutdown callbac
- Ensure that the audio subsystem is powered down cleanly when the syste
- shuts down by providing a shutdown operation. This ensures that all th
- components have been returned to an off state cleanly which should avoi
- audio issues from partially charged capacitors or noise on digital input
- if the system is restarted quickly
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Tested-by: Ben Dooks <ben-linux@fluff.org
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Make DAPM power sequence lists local variable
- They are now only accessed within dapm_power_widgets() so can be loca
- to that function
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Coalesce power updates for PGA
- Handle gain ramping for PGAs so we can coalesce their power updates too
- This is not ideal since we can't cope properly with gain ramping fo
- stereo paths but that was the case without coalescing and gain rampin
- is relatively infrequently used so the effects are limited
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Coalesce power updates for DAPM widgets with event
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Sort specialised mixers and muxes togethe
- The more flexible value muxes and named mixers don't need to be sorte
- differently from a power management point of view, they are differen
- only in terms of the control interface and not in terms of seqencin
- behaviour
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Coalesce register writes for DAPM sequence
- Reduce the number of register writes we need to set the power state fo
- a CODEC by coalescing updates to widgets with the same sequence order an
- same register into a single write
- This can be a noticable performance improvement with slow or heavil
- contended control buses, such as I2C controllers with a low cloc
- frequency, and is particularly noticable when resuming. It can als
- reduce the noticability of and pops and clicks by ensuring that lef
- and right channels are powered simultaneously if they are in the sam
- register
- Currently widgets that have events are not coalesced, including PGA
- which may use the volume ramping control
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Allow 32 bit registers for DAP
- Replace the remaining unsigned shorts with unsigned ints
- Tested with pcap2 codec (25 bits registers)
- Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out DAPM sequence executio
- Lump the list walk into a single function, and pull in the powe
- application too so we can do some further refactoring. Pure cod
- motion
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Sort DAPM power sequences while building list
- In the past the DAPM power sequencing was done by iterating over the lis
- of widgets once for each widget type and powering widgets of that type
- Instead of doing that do the sorting at the time we insert the widget
- into the lists of widgets to apply power changes to. This reduces th
- amount of computation required for seqencing still further, though th
- costs are generally dwarfed by the costs of the register write
- implementing them
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Apostrophe patro
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: Add debug trace for bias level transition
- A standard way of making sure we know when the bias level changes
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Integrate bias management with DAPM power managemen
- Rather than managing the bias level of the system based on if there i
- an active audio stream manage it based on there being an active DAP
- widget. This simplifies the code a little, moving the power handlin
- into one place, and improves audio performance for bypass paths when n
- playbacks or captures are active
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Make DAPM sysfs entries non-optiona
- sysfs is so standard these days there's no point
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Split DAPM power checks from sequencing of power change
- DAPM has always applied any changes to the power state of widgets as soo
- as it has determined that they are required. Instead of doing this stor
- all the changes that are required on lists of widgets to power up an
- down, then iterate over those lists and apply the changes. This change
- the sequence in which changes are implemented, doing all power down
- before power ups and always using the up/down sequences (previously the
- were only used when changes were due to DAC/ADC power events). The erro
- handling is also changed so that we continue attempting to power widget
- if some changes fail
- The main benefit of this is to allow future changes to do optimisation
- over the whole power sequence and to reduce the number of walks of th
- widget graph required to check the power status of widgets
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add power supply widget to DAP
- Many modern CODECs have shared resources on chip which must be enable
- for portions of the chip to work but which can be disabled at other time
- in order to achieve power savings. Examples of such resources includ
- power supplies and some internal clocks
- Since these widgets are dependencies for the audio path but do not carr
- audio signals they require slightly different handling to most widgets
- they do not contribute to the audio path and so should not be counted a
- either inputs or outputs during path walks
- Cases where one supply provides a supply for another will requir
- additional work. There is also room for more optimisation of the grap
- walking to avoid repeated checks for the same thing
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Make the DAPM power check an operation on the widge
- Rather than having switch statements at point of use make the DAP
- power check a member of the widget structure and set it when w
- instantiate the widget
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out DAPM power checks for DACs and ADC
- This also switches us to using a switch statement for the widget typ
- in dapm_power_widget()
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out generic widget power check
- This will form a basis for further power check refactoring: the overal
- goal of these changes is to allow us to check power separately t
- applying it, allowing improvements in the power sequencing algorithms
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Support DAPM events for DACs and ADC
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out application of power for generic widget
- This is simple code motion, intended to support future refactoring o
- the DAPM algorithms and (more immediately) the additon of events fo
- DACs and ADCs
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Display return code when failing to add a DAPM kcontro
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC FSI SH7724
- - ASoC: Add SuperH FSI driver support for ALS
- This driver is very simple
- It support playback only now
- This patch is tested by ms7724se board
- Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Freescale
- - ASoC: MPC5200: Support for buffer wrap aroun
- The code in psc_dma_bcom_enqueue_tx() didn't account for the fact tha
- s->runtime->control->appl_ptr can wrap around to the beginning of th
- buffer. This change fixes this problem
- Signed-off-by: John Bonesio <bones@secretlab.ca
- Acked-by: Grant Likely <grant.likely@secretlab.ca
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add missing DRV_NAME definitions for fsl/* driver
- Module builds are broken due to missing DRV_NAME fo
- efika-audio-fabric and pcm030-audio-fabric
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: MPC5200: Increase the delay time between reset
- Reset was failing with the original udelay(50) between the code i
- psc_ac97_cold_reset() and the call to psc_ac97_warm_reset(). Through testin
- it was found that a delay of 1ms was necessary for the cold_reset code t
- consistently complete successfully
- Signed-off-by: John Bonesio <bones@secretlab.ca
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: add locking to mpc5200-psc-ac97 drive
- AC97 bus register read/write hooks need to provide locking, but th
- mpc5200-psc-ac97 driver does not. This patch adds a mutex aroun
- the register access routines
- Signed-off-by: Grant Likely <grant.likely@secretlab.ca
- Acked-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix mpc5200-psc-ac97 to ensure the data ready bit is cleare
- When doing register reads, it is possible for there to be a stal
- data ready bit set which will cause subsequent reads to retur
- prematurely with incorrect data. This patch fixes the issues b
- ensuring stale data is cleared before starting another transaction
- Signed-off-by: Grant Likely <grant.likely@secretlab.ca
- Acked-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: remove BROKEN from Efika and pcm030 fabric driver
- The needed spin_event_timeout() macro is now merged in from th
- powerpc tree, so these drivers are no longer broken. This revert
- commit 0c0e09e21a9e7bc6ca54e06ef3d497255ca26383 (ASoC: Mark MPC520
- AC97 as BROKEN until PowerPC merge issues are resolved
- Tested against 2.6.31-rc1
- Signed-off-by: Grant Likely <grant.likely@secretlab.ca
- Acked-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: Fix typo in MPC5200 PSC AC97 driver Kconfi
- ALSA SoC drivers should be specify SND_SOC_AC97_BUS instead, not AC97_BUS
- Without SND_SOC_AC97_BUS defined, an AC97 device will not get correctl
- registered on the AC97 bus, which prevents thinks like the WM971
- touchscreen driver from getting probed
- Tested against 2.6.31-rc1
- Signed-off-by: Grant Likely <grant.likely@secretlab.ca
- Acked-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout(
- The function signature for spin_event_timeout() has changed in version V9
- Adjust the mpc5200 AC97 driver to use the new function
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Acked-by: Timur Tabi <timur@freescale.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Switch FSL SSI DAI over to symmetric_rate
- The effect of symmetric_constraints should provide a standard way t
- enforce the use of the same sample rate for both directions
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Acked-by: Timur Tabi <timur@freescale.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolve
- These drivers use spin_event_timeout() which is only present in th
- PowerPC tree at present and which is undergoing some API revision
- so temporarily mark them as BROKEN until these issues are sorte
- out
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fabric bindings for STAC9766 on the Efik
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Support for AC97 on Phytec pmc030 base board
- A wm9712 AC97 codec is used
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: AC97 driver for mpc520
- I've implemented retries for when the AC97 hardware doesn't reset o
- first try. About 10% of the time both the Efika and pcm030 AC97 codec
- don't reset on first try and need to be poked multiple times. Failur
- is indicated by not having the link clock start ticking. Every once i
- a while even five pokes won't get the link started and I have to powe
- cycle
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Main rewite of the mpc5200 audio DMA cod
- Rewrite the mpc5200 audio DMA code to support both I2S and AC97
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Acked-by: Grant Likely <grant.likely@secretlab.ca
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Rename the PSC functions to DM
- Rename the functions in the mpc5200 DMA file from i2s based names to dm
- ones to reflect the file they are in
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Acked-by: Grant Likely <grant.likely@secretlab.ca
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Basic split of mpc5200 DMA code out of mpc5200_psc_i2
- Basic split of mpc5200 DMA code out from i2s into a standalone file
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Acked-by: Grant Likely <grant.likely@secretlab.ca
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: use dev_set_drvdat
- Eliminate direct accesses to the driver_data field
- cf 82ab13b26f15f49be45f15ccc96bfa0b81dfd01
- The semantic patch that makes this change is as follows
- (http://www.emn.fr/x-info/coccinelle/
- // <smpl
- @
- struct device *dev
- expression E
- type T
- @
- - dev->driver_data = (T)
- + dev_set_drvdata(dev, E
- @
- struct device *dev
- type T
- @
- - (T)dev->driver_dat
- + dev_get_drvdata(dev
- // </smpl
- Signed-off-by: Julia Lawall <julia@diku.dk
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove BROKEN from mpc5200 kconfi
- The regression was fixed by commi
- 3e5b50165fd0be080044586f43fcdd460ed27610, so no need to mark thi
- driver as BROKEN
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: Set the MPC5200 i2s driver to BROKEN status
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Acked-by: Grant Likely <grant.likely@secretlab.ca
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
SoC Layer
- - Fix build of soc-core.c with older kernel
- Now it's using dev_pm_ops, which was added recently
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ASoC: fix I2C build error
- Fix soc build errors when I2C is built as a loadable module
- (.text+0x5d26b): undefined reference to `i2c_master_send
- soc-cache.c:(.text+0x5d32d): undefined reference to `i2c_transfer
- Signed-off-by: Randy Dunlap <randy.dunlap@oracle.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add DAPM widget power decision debugfs file
- Currently when built with DEBUG DAPM will dump information abou
- the power state decisions it is taking for each widget to dmesg
- This isn't an ideal way of getting the information - it require
- a kernel build to turn it on and off and for large hub CODECs th
- volume of information is so large as to be illegible. When th
- output goes to the console it can also cause a noticable impac
- on performance simply to print it out
- Improve the situation by adding a dapm directory to our debugf
- tree containing a file per widget with the same information i
- it. This still requires a decision to build with debugfs suppor
- but is easier to navigate and much less intrusive
- In addition to the previously displayed information active stream
- are also shown in these files
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add ak4642/ak4643 codec suppor
- This is very simple driver for ALS
- It supprt headphone output and stereo input onl
- This patch is tested by ms7724s
- Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Hook i.MX into buil
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out shared code from WM899
- The WM8993 analogue control is shared with other devices in the sam
- product line. Since this is a very substantial proportion of th
- driver move the definitions of these controls into a new wm_hubs modul
- which allows them to be shared between the two
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Minor cleanups to AD1938 drive
- - Build in SND_SOC_ALL_CODECS
- - Remove null suspend/resume stuff
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: new ad1836 codec driver based on aso
- There has been an ad1836 driver in sound/blackfin based on traditional alsa
- The new driver is based on asoc. The architecture of ad1836 codec driver i
- very much like ad1938
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Define more formats for the AC97 CODEC
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: change set_tdm_slot api to allow slot_width override
- Extend set_tdm_slot to allow the user to arbitrarily set the frame widt
- and active TX/RX slots
- Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.
- still doesn't handle the slot_width override
- While being there, correct an incorrect use of SlotsPerFrm(7) use i
- bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) )
- (this series is meant for Mark's for-2.6.32 branch
- Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8776 CODEC drive
- The WM8776 is a high performance, stereo audio CODEC with five channe
- input selector. The WM8776 is ideal for surround sound processin
- applications for home hi-fi, DVD-RW and other audio visual equipment
- This driver implements support for most WM8776 features - currently th
- ADC automatic level control/limiter functionality is omitted
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Factor out I2C 8 bit address 16 bit data I/
- As part of this refactoring the type of the CODEC hw_read operatio
- is changed to match the regular read operation
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add I/O control bus information to factored out cache setu
- While writes tend to be able to use a fairly bus independant format t
- do the writes reads are all bus specific. To allow us to factor ou
- this code include the bus type as a parameter when setting up th
- cache
- Initially just use this to factor out hw_write_t for I2C
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: jack: Fix race in snd_soc_jack_add_gpio
- The irq can fire as soon as it has been requested, thus all fields accesse
- from within the irq handler must be initialized prior to requesting the irq
- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Allow CODECs to flag invalid register
- This helps CODECs with sparse register maps work better with th
- register cache display interface
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Begin to factor out register cache I/O function
- A lot of CODECs share the same register data formats and therefor
- replicate the code to manage access to and caching of the registe
- map. In order to reduce code duplication centralised versions o
- this code will be introduced with drivers able to configure the us
- of the common code by calling the new snd_soc_codec_set_cache_io(
- API call during startup
- As an initial user the 7 bit address/9 bit data format used by man
- Wolfson devices is supported for write only CODECs and the driver
- with straightforward register cache implementations are converted t
- use it
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8974 CODEC drive
- The WM8974 is a low power, high quality mono CODEC designed for portabl
- applications such as digital still cameras or digital voice recorders
- This driver was originally written by Graeme Gregory and Liam Girdwoo
- and has since been maintained by myself with some updates contributed b
- Brett Saunders and Javier Martin
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Jack handling enhancements as suggested by subsystem maintaine
- The patch adds a few small enhancements to the ASoC jack handling, a
- suggested by Mark in his comments to my Amstrad Delta driver, and a few fixe
- for related bugs found while learning Mark's code and testing results
- Enhancements
- 1. Update status of an ASoC jack while associating it with new gpios
- 2. Really update DAPM pins while associating them with an ASoC jack
- 3. Export ASoC jack gpios over gpiolib sysfs for diagnostic purposes
- Fixes
- 1. Apply mask on jack status report before using it, just for case
- 2. While updating jack associated DAPM pins, use full resulting jack status
- not the status report passed as an argument
- Created and tested on linux-2.6.31-rc
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: Allow passing platform_data to devices attached to AC97 bu
- This patch allows passing platform_data to devices attached to AC97 bu
- (like touchscreens, battery measurement chips ...)
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add support for Conexant CX20442-11 voice modem code
- This patch adds support for Conexant CX20442-11 voice modem codec, suitabl
- for use by the ASoC board driver for Amstrad E3 (Delta) videophone. Relate
- sound card driver will follow
- This codec is an optional part of the Conexant SmartV three chip modem design
- As such, documentation for its proprietary digital audio interface is no
- available. However, on Amstrad Delta board, thanks to Mark Underwood wh
- created an initial, omap-alsa based sound driver a few years ago[1], the code
- has been discovered to be accessible not only from the modem side, but als
- over the OMAP McBSP based CPU DAI. Thus, the driver can be used by any soun
- card that can access the codec DAI directly. The DAI configuration parameter
- (sample rate and format, number of channels) has been selected out empiricall
- for best user experience
- The codec analogue interface consists of two pairs of analogue I/O pins
- speakerphone interface or telephone handset/headset interface. Furthermore, i
- seams to provide two operation modes for speakerphone I/O: standard an
- advanced, with automatic gain control and echo cancelation. Even if the code
- control interface is unknown and not available, all those interfaces and mode
- can be selected over the modem chip using V.253 commands. The driver is abl
- to issue necessary commands over a suitable hw_write function if provided by
- sound card driver. Otherwise, the codec can be controlled over the modem fro
- userspace while inactive
- Even if nothig is known about the codec internal power managemen
- capabilities, DAPM widgets has been used to model the codec audio map
- Automatically performed powering up/down of those virtual widgets results i
- corresponding V.253 commands being issued
- Some driver features/oddities may be board specific, but I have no way t
- verify that with any board other than Amstrad Delta
- [1] http://www.earth.li/pipermail/e3-hacking/2006-April/000481.htm
- Created and tested against linux-2.6.31-rc3
- Applies and works with linux-omap-2.6 commi
- 7c5cb7862d32cb344be7831d466535d5255e35ac as well
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: new ad1938 codec driver based on aso
- Signed-off-by: Barry Song <21cnbao@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: MAX9877: add MAX9877 amp drive
- The MAX9877 combines a high-efficiency Class D audio power amplifie
- with a stereo Class AB capacitor-less DirectDrive headphone amplifier
- The max9877_add_controls() is called to register the MAX9877 specifi
- controls on machine specific init() of the machine driver
- The datasheet for the MAX9877 can find at the following url
- http://datasheets.maxim-ic.com/en/ds/MAX9877.pd
- [Slight edit to sort the ALL_CODECS entries -- broonie.
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: add SOC_DOUBLE_R_EXT_TLV control typ
- This is a macro for double controls with special callback function an
- TLV. The SOC_DOUBLE_R_EXT_TLV needs two registers and one shift fo
- double controls
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: add SOC_DOUBLE_EXT_TLV control typ
- This is a macro for double controls with special callback function an
- TLV. The SOC_DOUBLE_EXT_TLV needs one register and two shifts for doubl
- controls
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: fixes multiple typos in comments, no functional chang
- Signed-off-by: Peter Meerwald <pmeerw@pmeerw.net
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8993 CODEC drive
- The WM8993 is a highly integrated ultra-low power hi-fi CODEC designe
- for portable devices such as multimedia phones
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add CODEC volatile register operatio
- Add a volatile_register() operation to the CODEC structure providing
- standard operation to query if a register is volatile. This will be use
- to factor out the register cache I/O operations for the CODECs
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8523 CODEC drive
- The WM8523 is a high performance stereo DAC with integral charg
- pump providing 2Vrms line driver outputs using a single 3.3V powe
- supply rail
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Convert to dev_pm_op
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add a shutdown callbac
- Ensure that the audio subsystem is powered down cleanly when the syste
- shuts down by providing a shutdown operation. This ensures that all th
- components have been returned to an off state cleanly which should avoi
- audio issues from partially charged capacitors or noise on digital input
- if the system is restarted quickly
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Tested-by: Ben Dooks <ben-linux@fluff.org
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add stub suspend and resume calls for ASoC subdevice
- Now that ASoC subdevices can be regular devices they can have norma
- suspend and resume calls from their buses. However, suspending the
- individually is not desirable since this can lead to problems such a
- pops and clicks from devices being suspended with their signals bein
- amplified or clocks being stopped suddenly
- This will be resolved by having the normal device model suspend an
- resume calls call into ASoC which will suspend the entire card while an
- of its components are suspended. At present this is not yet implemente
- but in order to aid the transition of drivers to the standard devic
- model this patch adds API calls for the notifications
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8961 drive
- The WM8961 is a low power, high quality stereo CODEC designed fo
- portable digital applications with headphone and stereo class D speake
- drivers
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Make DAPM power sequence lists local variable
- They are now only accessed within dapm_power_widgets() so can be loca
- to that function
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Allow 32 bit registers for DAP
- Replace the remaining unsigned shorts with unsigned ints
- Tested with pcap2 codec (25 bits registers)
- Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Instantiate any forgotten DAPM widget
- With the recent changes to the DAPM power checks it has become importan
- to explicitly instantiate all widgets but some drivers were forgettin
- to do that. Since everything needs to do it add a call to instantiat
- them immediately before the card registration - it does no harm when i
- is called repeatedly and saves work in drivers
- Tested-by: pHilipp Zabel <philipp.zabel@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: fix NULL pointer dereference in soc_suspend(
- In case the initalization of an soc_device failed, there is no code
- associated with it. soc_suspend() will still dereference the pointe
- and cause an Ooops when entering the sleep mode
- This happens on our board with a multi-target kernel image when boote
- on a machine without audio circuits
- This patch makes the code bail out very early in this special case
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add dummy S/PDIF codec suppor
- McASP on DM646x can operate in DIT (S/PDIF) where no codec is needed
- This patch provides stub codec that can be used in these configurations
- On DM646x EVM the McASP1 is connected to the S/PDIF out
- Signed-off-by: Steve Chen <schen@mvista.com
- Signed-off-by: Pavel Kiryukhin <pkiryukhin@ru.mvista.com
- Signed-off-by: Naresh Medisetty <naresh@ti.com
- Signed-off-by: Chaithrika U S <chaithrika@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Codec for STAC9766 used on the Efik
- Datasheet: http://www.idt.com/products/getDoc.cfm?docID=1313400
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: WM9081 mono DAC with integrated 2.6W class AB/D amplifier drive
- The WM9081 is designed to provide high power output at low distortio
- levels in space-constrained portable applications
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - AsoC: Make snd_soc_read() and snd_soc_write() function
- Should be no impact on the generated code but it helps the compile
- print clearer messages
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add TXx9 AC link controller driver (v3
- This patch adds support for the integrated ACLC of the TXx9 family
- Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Integrate bias management with DAPM power managemen
- Rather than managing the bias level of the system based on if there i
- an active audio stream manage it based on there being an active DAP
- widget. This simplifies the code a little, moving the power handlin
- into one place, and improves audio performance for bypass paths when n
- playbacks or captures are active
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Split DAPM power checks from sequencing of power change
- DAPM has always applied any changes to the power state of widgets as soo
- as it has determined that they are required. Instead of doing this stor
- all the changes that are required on lists of widgets to power up an
- down, then iterate over those lists and apply the changes. This change
- the sequence in which changes are implemented, doing all power down
- before power ups and always using the up/down sequences (previously the
- were only used when changes were due to DAC/ADC power events). The erro
- handling is also changed so that we continue attempting to power widget
- if some changes fail
- The main benefit of this is to allow future changes to do optimisation
- over the whole power sequence and to reduce the number of walks of th
- widget graph required to check the power status of widgets
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add SNDRV_PCM_FMTBIT_S32_BE as a valid AC97 forma
- Signed-off-by: Jon Smirl <jonsmirl@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Fix up CODEC DAI formats for big endian CPU
- ASoC uses the standard ALSA data format definitions to specify the wir
- format used between the CPU and CODEC. Since the ALSA data formats al
- include the endianess of the data but this information is not relevan
- by the time the data has been encoded onto the serial link to the CODE
- this means that either all the CODEC drivers need to declare both big an
- little endian variants or the core needs to fix up the format constraint
- specified by CODEC drivers
- For now take the latter approach - this will need to be revisited if an
- CODECs are endianness dependant
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove redundant codec pointer from DAI
- The DAI structure has two pointers to the codec, one in the body of th
- DAI and one in a union for a parent pointer. Drop the parent pointe
- version
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Remove unused DAI format define
- The defines for TDM and synchronous clocks are not used - they ar
- mostly a legacy of the automatic clocking configuration. TDM wil
- require configuration of the number of timeslots and which ones to us
- so can't be fit into the DAI format and synchronous mode is handled b
- symmetric_rates (and needs to be done by constraints rather than whe
- the DAI format is being configured)
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Use a shared define for AC97 CODEC data format
- The AC97 wire format is completely fixed so CODECs don't have any choic
- about the formats they accept but controllers accept a variety of dat
- formats and render them down onto the bus. Have a shared define so al
- the CODEC drivers will interoperate with any of our controller drivers
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: ASoC WM8940 Drive
- Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: add SOC_DOUBLE_EXT macr
- Add a macro for double controls with special callback functions
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Volume controls are never of boolean typ
- Some limited volume controls (mostly simple attenuations) have only tw
- settings so the ASoC info functions misreport them as booleans. Sinc
- we currently have no better information check for " Volume" in th
- control name and always report any controls matching as being integer
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Check we have DAI ops when calling via accessor function
- Also make sure we're checking for the right operation while we're here
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8960 CODEC drive
- The WM8960 is a low power, high quality stereo codec designed fo
- portable digital audio applications
- Stereo class D speaker drivers provide 1W per channel into 8W loads
- Guaranteed low leakage, excellent PSRR and pop/click suppressio
- mechanisms enable direct battery connection for the speaker supply
- The device also integrates a complete microphone interface and a stere
- headphone driver. External component requirements are drasticall
- reduced as no separate microphone, speaker or headphone amplifiers ar
- required. Advanced on-chip digital signal processing performs automati
- level control for the microphone or line input
- Stereo 24-bit sigma-delta ADCs and DACs are used with low powe
- over-sampling digital interpolation and decimation filters and
- flexible digital audio interface
- The master clock can be input directly or generated internally by a
- onboard PLL, supporting most commonly-used clocking schemes
- This driver was originally written by Liam Girdwood, with substantia
- subsequent additions and updates for feature completeness and changes i
- the ASoC framework from me
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add WM8988 CODEC drive
- The WM8988 is a low power, high quality stereo CODEC designed fo
- portable digital audio applications
- The device integrates complete interfaces to 2 stereo headphone or lin
- out ports. External component requirements are drastically reduced as n
- separate headphone amplifiers are required. Advanced on-chip digita
- signal processing performs graphic equaliser, 3-D sound enhancement an
- automatic level control for the microphone or line input
- The WM8988 can operate as a master or a slave, with various master cloc
- frequencies including 12 or 24MHz for USB devices, or standard 256f
- rates like 12.288MHz and 24.576MHz. Different audio sample rates such a
- 96kHz, 48kHz, 44.1kHz are generated directly from the master cloc
- without the need for an external PLL
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Provide core support for symmetric sample rate
- Many devices require symmetric configurations of capture and playbac
- data formats, often due to shared clocking but sometimes also due t
- other shared playback and record configuration in the device. Star
- providing core support for this by allowing the DAIs or the machin
- to specify that the sample rates used should be kept symmetric
- A flag symmetric_rates is provided in the snd_soc_dai an
- snd_soc_dai_link structures. If this is set in either of the DAIs or i
- the machine then a constraint will be applied when a stream is alread
- open preventing any changes in sample rate
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: soc-core: fix crash when removing not instantiated car
- If the card was not instantiated in snd_soc_instantiate_card, callin
- soc-remove will crash because some of codec, cpu_dai and card .remov
- methods are called twice
- Fix this by returning from soc_remove immediately
- Signed-off-by: Mike Rapoport <mike@compulab.co.il
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: Add driver for s6000 I2S interfac
- This patch adds a driver for the I2S interface found on Stretch s600
- family processors
- Signed-off-by: Daniel Glöckner <dg@emlix.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC PXA2xx Corgi
- - [ARM] pxa: register wm8731 explicitly for corgi and poodl
- The wm8731 driver has been converted to register using the standard I2
- device registration mechanism so we can now do the registration in th
- arch/arm code as normal
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Eric Miao <eric.y.miao@gmail.com
SoC PXA2xx EM-X270
- - ASoC: em-x270: make the driver support also eXeda and CM-X300 machine
- Signed-off-by: Mike Rapoport <mike@compulab.co.il
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC PXA2xx Palm T|X
- - ASoC: Switch palm27x-asoc to jack detection ap
- This patch removes the old method of jack detection from palm27x-aso
- driver and adds jack detection api. It also removes some other (now
- useless stuff from the driver and corrects pin configuration for th
- codec
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - [ARM] pxa/palm: Add Palm27x aSoC driver to PalmTE
- Signed-off-by: Marek Vasut <marek.vasut@gmail.com
- Signed-off-by: Eric Miao <eric.miao@marvell.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC PXA2xx Poodle
- - [ARM] pxa: register wm8731 explicitly for corgi and poodl
- The wm8731 driver has been converted to register using the standard I2
- device registration mechanism so we can now do the registration in th
- arch/arm code as normal
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Eric Miao <eric.y.miao@gmail.com
SoC S6000
- - ASoC: tlv320aic3x: fixup board device change
- Fixup the device changes by modifying the files that we just removed th
- explicit device creation from with i2c_register_board_info() until thi
- can be moved into the relevant board files
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: tlv320aic3x: Change to use device mode
- The tlv320aic3x driver managed its own i2c device, instead of an extan
- one created by the board support code. Change the code to make it so tha
- the driver binds to an extant (in this case i2c) device
- Add explict tlv320aic33 as well as tlv320aic3x to the supported devic
- table and remove the old driver bindings from the users of this code
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: correct s6000 I2S clock polarit
- According to the data sheet data is clocked out on the falling edg
- and latched on the rising edge of the bit clock. While the left sampl
- is transmitted the word clock line is low
- Signed-off-by: Daniel Glöckner <dg@emlix.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: s6105 IP camera machine specific ASoC cod
- This patch adds machine specific code for the audio part of the Stretc
- s6105 IP camera reference design
- The device uses the tlv320aic31(01) codec to generate the clock fo
- both I2S ports of the soc. While the master clock is generated by
- configurable PLL chip, the code assumes the factory default settings
- An additional kcontrol has been added to handle the special routing o
- the board, connecting both HPLCOM and HPROUT to the same pin of the audi
- jack. One of these should always be switched off
- Signed-off-by: Daniel Glöckner <dg@emlix.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add driver for s6000 I2S interfac
- This patch adds a driver for the I2S interface found on Stretch s600
- family processors
- Signed-off-by: Daniel Glöckner <dg@emlix.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC SH7760 AC97
- - ASoC: Add FSI-AK4642 sound support for Super
- This patch is tested by ms7724s
- Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add SuperH FSI driver support for ALS
- This driver is very simple
- It support playback only now
- This patch is tested by ms7724se board
- Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
SoC Texas Instruments OMAP
- - sound: TTY/ASoC: Rename N_AMSDELTA line discipline to N_V25
- The patch changes the line discipline name registered in include/linux/tty.
- and updates the ams-delta machine driver to use it
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: SDP3430: Fix TWL GPIO6 pin mux reques
- Fix the write to PMBR1 register through I2C. Also, the constant whic
- holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. Thi
- name is based on TRM to avoid confusion
- Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_sto
- Functionality of functions omap_mcbsp_xmit_enable and omap_mcbsp_recv_enabl
- can be merged into omap_mcbsp_start and omap_mcbsp_stop since API o
- those omap_mcbsp_start and omap_mcbsp_stop was changed recently allowin
- to start and stop individually the transmitter and receiver
- This cleans up the code in arch/arm/plat-omap/mcbsp.c and i
- sound/soc/omap/omap-mcbsp.c which was the only user for those remove
- functions
- Signed-off-by: Jarkko Nikula <jhnikula@gmail.com
- Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com
- Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DA
- Commit ca6e2ce08679c094878d7f39a0349a7db1d13675 is setting up few XCCR an
- RCCR bits for I2S and DPS_A formats. Part of the bits are already se
- for all formats and I believe that XDISABLE and RDISABLE bits ar
- format independent
- As XCCR and RCCR are found only from OMAP2430 and OMAP34xx, I move setu
- of XDISABLE and RDISABLE to where those cpu's are tested and remove forma
- dependent part for simplicity
- Signed-off-by: Jarkko Nikula <jhnikula@gmail.com
- Acked-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com
- Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: tlv320aic3x: fixup board device change
- Fixup the device changes by modifying the files that we just removed th
- explicit device creation from with i2c_register_board_info() until thi
- can be moved into the relevant board files
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: tlv320aic3x: Change to use device mode
- The tlv320aic3x driver managed its own i2c device, instead of an extan
- one created by the board support code. Change the code to make it so tha
- the driver binds to an extant (in this case i2c) device
- Add explict tlv320aic33 as well as tlv320aic3x to the supported devic
- table and remove the old driver bindings from the users of this code
- Signed-off-by: Ben Dooks <ben@simtec.co.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Use DMA operating mode of McBS
- Configures DMA sync mode depending on McBSP operating mode value
- The value is configurable by McBSP instance. So, dependin
- on McBSP operating mode, the DMA sync mode is passed fro
- omap-mcbsp to omap-pcm. Besides that, it also configure
- McBSP threshold value depending on which McBSP mode is activated
- Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com
- Acked-by: Jarkko Nikula <jhnikula@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Use McBSP threshold to playback and captur
- This patch changes the way DMA is done in omap-pcm.
- in order to reduce power consumption. There is no nee
- to have so much SW control in order to have DMA in idl
- state during audio streaming. Configuring McBSP threshold valu
- and DMA to FRAME_SYNC are sufficient
- Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com
- Acked-by: Jarkko Nikula <jhnikula@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Always syncronize audio transfers on frame
- All these steps are required for ASoC to behave correctly
- rccr and xccr are format dependent, for example TDM audi
- has different values than I2S or DSP_A. Also th
- omap_mcbsp_xmit_enable and/or omap_mcbsp_recv_enable mus
- be called right after the DMA has started
- This provides no longer L and R channels switching at random
- Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com
- Acked-by: Jarkko Nikula <jhnikula@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Add runtime check for RFIG and XFI
- This is, no RFIG or XFIG (Not defined in 34xx), correc
- initiliazation of rccr and xccr
- Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com
- Acked-by: Jarkko Nikula <jhnikula@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Make DMA 64 aligne
- Align DMA address to DMA burst transaction sizes
- Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com
- Acked-by: Jarkko Nikula <jhnikula@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Enable DMA burst mod
- Improve DMA transfers by enabling Burst transaction
- Signed-off-by: Eduardo Valentin <eduardo.valentin@nokia.com
- Acked-by: Jarkko Nikula <jhnikula@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Enhance OMAP1510 DMA progress software counte
- Enhance period_index accuracy, particularly just before buffer rewind, b
- making use of DMA interrupt status flags in addition to simply counting u
- interrupts
- Created against linux-2.6.31-rc5
- Tested on Amstrad Delta
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Acked-by: Jarkko Nikula <jhnikula@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Make use of DMA channel self linking on OMAP151
- Use newly implemented DMA channel self linking on OMAP1510 like on other OMA
- models. Remove unnecessary DMA transfer restart from interrupt handle
- routine
- The interrupt routine used to maintain a period index, originally needed fo
- counting up periods up to a full buffer in order to restart the DMA transfer
- For some time, this counter is also used as a replacement for hardware DM
- progress counter that has been found unusable on OMAP1510 in case of playback
- Thus, the period index calculation cannot be omitted completely. However, th
- accuracy of this counter can still suffer from missing DMA interrupts
- In order to work correctly, it requires patch 1 from this series also applied
- [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP151
- Created against linux-2.6.31-rc5
- Tested on Amstrad Delta
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Acked-by: Jarkko Nikula <jhnikula@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/sto
- Simultaneous audio playback and capture on OMAP1510 can cause that secon
- stream is stalled if there is enough delay between startup of the audi
- streams
- Current implementation of the omap_mcbsp_start is starting both transmitte
- and receiver at the same time and it is called only for firstly starte
- audio stream from the OMAP McBSP based ASoC DAI driver
- Since DMA request lines on OMAP1510 are edge sensitive, the DMA request i
- missed if there is no DMA transfer set up at that time when the first wor
- after McBSP startup is transmitted. The problem hasn't noted before sinc
- later OMAPs are using level sensitive DMA request lines
- Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop b
- allowing to start and stop individually McBSP transmitter and receive
- logics. Then call those functions individually for both audio playbac
- and capture streams. This ensures that DMA transfer is setup befor
- transmitter or receiver is started
- Thanks to Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> for detailed proble
- analysis and Peter Ujfalusi <peter.ujfalusi@nokia.com> for info about DM
- request line behavior differences between the OMAP generations
- Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Jarkko Nikula <jhnikula@gmail.com
- Acked-by: Tony Lindgren <tony@atomide.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: add support for Amstrad E3 (Delta) machin
- This patch adds machine support for Amstrad E3 (Delta) videophone to ASoC
- Created and tested against linux-2.6.31-rc3
- Applies and works with linux-omap-2.6 commi
- 7c5cb7862d32cb344be7831d466535d5255e35ac as well
- Depends on
- 1) latest version of the CX20442 codec driver that exposes v253_op
- structure[1]
- 2) patch 2/3 form this series: TTY: Add definition of a new lin
- discipline required by Amstrad E3 (Delta) ASoC driver[2]
- CPU DAI parameters best matching the codec DAI has been selected ou
- empirically for best user experience
- Board specific audio function control (with related DAPM widgets) has bee
- modeled after empirically discovered codec capabilities
- Unlike other ASoC machine drivers, this one makes use of a codec provided lin
- discipline that is required for talking to a modem chip that can control th
- codec behavoiur. As the line discipline operations must call board specifi
- bits as well, the machine driver registers its own line discipline ops, no
- the codec provided, and then calls those codec provided from inside its ow
- callbacks
- If some kind of a glue, like a bus over a tty, exsited that could help i
- runtime detection of a modem (bus adapter) over a more generic line disciplin
- (bus driver)[3], the line discipline code could be probably designed in
- more generic way
- In order to work at all, this driver requires a working McBSP1. On OMAP151
- based machines (not sure if other OMAP1 variants as well), where McBSP1 is
- DSP public peripheral, that means the kernel must provide basic DSP support
- ie. omap_dsp_init(), in order to power up the DSP. This used to be included i
- linux-omap-2.6 tree up to commit 2512fd29db4eb09e82d182596304c7aaf76d2c5c
- Without that, the driver would not work, ie. not shift in/out any bits ove
- the CPU DAI[4]. This limitation is not board, but CPU specific, and may appl
- to other code that makes use of McBSP1/McBSP3 on affected machines. I provid
- an extra patch (4/3) as a temporary solution
- To work correctly in playback mode, this driver requires my prevoiusl
- submitted patch that corrects pcm pointer calculation for OMAP1510 base
- machines[5] (already included in linux-2.6.31-rc3)
- To support codec controls, this driver requires my previously submitted patc
- that adds support for modem found on Amstrad Delta[6]
- [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-July/019780.htm
- [2] http://www.spinics.net/lists/linux-serial/msg01862.htm
- [3] http://www.spinics.net/lists/linux-serial/msg01856.htm
- [4] http://www.spinics.net/lists/linux-omap/msg15114.htm
- [5] http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018950.htm
- [6] http://www.spinics.net/lists/linux-omap/msg15432.htm
- Credits to
- Mark Underwood - for his initial, omap-alsa based sound driver fo
- this machine
- Mark Brown - for his help, patience and excellent subsytem maintainer support
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Staticise pcm creation function of omap-pc
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO
- Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO
- line, controlled by register INTBR_PMBR1. Machine driver takes car
- of enabling gpio line through i2c and codec driver manipulates th
- line during headset ramp up/down sequence
- Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Zoom2: Update twl4030_setup_data parameter
- Add support for EXTMUTE in Zoom2 machine driver. This is necessar
- to further reduce pop noise problem. Signal EXTMUTE is connected t
- signal GPIO 153 in Zoom2 board
- In addition, change ramp delay value to 3 (218/161/109 ms). Wit
- previous ramp delay value, pop noise was louder. With a longer valu
- the beep tone can be observed
- Signed-off-by: Jorge Eduardo Candelaria <x0107209@ti.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Fix voice interface clock master
- Voice interface of twl4030 codec supports: CBM_CFM an
- CBS_CFS. It doesn't support CBS_CFM
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-By: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Zoom2: Add machine driver for Zoom2 boar
- Add support for Zoom2 board. Zoom2 machine drive
- connects both codec DAIs (audio and voice) to omap
- McBSP ports in the following way
- HiFi <-> McBSP
- Voice <-> McBSP
- The zoom2 driver has the following DAPM widgets
- * Ext Mic: MAINMIC, SUBMIC (with bias
- * Ext Spk: HFL, HF
- * Headset Stereophone: HSOL, HSO
- * Headset Mic: HSMIC (with bias
- * Aux In: AUXL, AUX
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: fix OMAP1510 broken PCM pointer callbac
- This patch tries to work around the problem of broken OMAP1510 PCM playbac
- pointer calculation by replacing DMA function call that incorrectly tries t
- read the value form DMA hardware with a value computed locally from a
- already maintained variable omap_runtime_data.period_index
- Tested on OMAP5910 based Amstrad Delta (E3) using work in progress ASo
- driver
- Based on linux-2.6-asoc.git v2.6.31-rc1
- Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl
- Acked-by: Jarkko Nikula <jhnikula@gmail.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: SDP4030: Use the twl4030_setup_data for headset pop-remova
- With this patch the initial headset pop-removal related values ar
- configured for the twl4030 codec (ramp delay and sysclk)
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: SDP3430: Connect twl4030 voice DAI to McBSP
- Connect twl4030 voice DAI to McBSP3 in sdp3430 machine driver
- Voice DAI init function enables corresponding interface b
- writting directly to VOICE_IF codec register
- Signed-off-by: Misael Lopez Cruz <x0052729@ti.com
- Acked-by: Peter Ujflausi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Added OMAP3 EVM support in ASoC
- Resending the patch after fixing the minor issues
- Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: Beagle: Add support for 4 channe
- This patch adds support for the four channel TDM mod
- on Beagle board
- Depending on the channel count, the interface needs to b
- configured differently (I2S for stereo DSP_A for four channels
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Add 4 channel support to mcbs
- Add 4 channel support to omap-mcbsp
- This mode is going to be used by the twl4030 codec, when i
- is configured in Option1 mode
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Add checking to detect bufferless pcm
- Add checking in hw_params and prepare to detect bufferless pcms(i.e. B
- <--> codec)
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: TWL4030: Add support Voice DA
- Add Voice DAI to support the PCM voice interface of the twl4030 codec
- The PCM voice interface can be used with 8-kHz(voice narrowband) o
- 16-kHz(voice wideband) sampling rates, and 16bits, and mono RX and mon
- TX or stereo TX
- The PCM voice interface has two mode
- - PCM mode1 : This uses the normal FS polarity and the rising edge o
- the clock signal
- - PCM mode2 : This uses the FS polarity inverted and the falling edg
- of the clock signal
- If the system master clock is not 26MHz or the twl4030 codec mode is no
- option2, the voice PCM interface is not available
- Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com
- Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Add DSP_A mode support for mcbs
- DSP_A mode is similar to the DSP_B, but the MSB is delayed wit
- one bclk (appears after the FS pulse and not under it)
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: OMAP: Use single-phase for DSP mod
- Use single-phase mode for the DSP mode and keep the dual phas
- mode for the I2S mode
- The mono (1 channel) mode already used single phase mode
- now it is more cleaner. There is no need to configure th
- second phase, when the single phase is used
- Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com
- Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: n810: replace BUG() with BUG_ON(
- Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Soc PXA2xx Imote 2
- - ASoC: IMote2 ASoC Suppor
- This patch adds the ASoC side of the board support for the Crossbo
- IMB400 daughter board
- Thanks to Crossbow for considerable assistance
- Signed-off-by: Jonathan Cameron <jic23@cam.ac.uk
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Soc PXA2xx Magician
- - ASoC: change set_tdm_slot api to allow slot_width override
- Extend set_tdm_slot to allow the user to arbitrarily set the frame widt
- and active TX/RX slots
- Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.
- still doesn't handle the slot_width override
- While being there, correct an incorrect use of SlotsPerFrm(7) use i
- bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) )
- (this series is meant for Mark's for-2.6.32 branch
- Signed-off-by: Daniel Ribeiro <drwyrm@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: UDA1380: refactor device registratio
- This patch mostly follows commit 5998102b9095fdb7c67755812038612afea315c
- "ASoC: Refactor WM8731 device registration" to make UDA1380 use standar
- device instantiation. Similarly, the I2C device registration temporaril
- moves into the magician machine driver before it will find its fina
- resting place in the board file
- At the same time, platform specific configuration is moved to platform dat
- and common power/reset GPIO handling moves into the codec driver
- Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ASoC: magician: fix PXA SSP clock polarit
- Follow-up fix needed since "ASoC: pxa-ssp.c fix clock/frame invert"
- Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com
- Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com
- - ASoC: Optimize switch/case in magician.
- Use default to optimize the switch/case in magicial_playback_hw_params()
- which also fixes the compile warnings below
- sound/soc/pxa/magician.c:89: warning: 'acds' may be used uninitialized in this functio
- sound/soc/pxa/magician.c:89: warning: 'acps' may be used uninitialized in this functio
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
USB
- - ALSA: snd_usb_caiaq: add support for Audio2D
- This adds support for Native Instrument's freshly announced Audio2D
- sound device hardware. Version number bumped to 1.3.19
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
USB USX2Y
- - Remove multiple KERN_ prefixes from printk format
- Commit 5fd29d6ccbc98884569d6f3105aeca70858b3e0f ("printk: clean u
- handling of log-levels and newlines") changed printk semantics. print
- lines with multiple KERN_<level> prefixes are no longer emitted a
- before the patch
- <level> is now included in the output on each additional use
- Remove all uses of multiple KERN_<level>s in formats
- Signed-off-by: Joe Perches <joe@perches.com
- Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org
- - ALSA: usx2y - reparent sound devic
- Fix the parent device to be the USB interface, not the USB device
- A similiar commit like 563c2bf59d392357bcc1d99642933cc88c687964
- Signed-off-by: Takashi Iwai <tiwai@suse.de
USB caiaq
- - Clean up useless files and fix .gitignore for caia
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: snd_usb_caiaq: add support for Audio2D
- This adds support for Native Instrument's freshly announced Audio2D
- sound device hardware. Version number bumped to 1.3.19
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: snd_usb_caiaq: reparent sound devic
- The sound device instance needs to be a child of the USB interface, no
- the USB device. Newer udev versions pay attention to that
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Reported-by: Lennart Poettering <lennart@poettering.net
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: snd_usb_caiaq: fix legacy input streamin
- Seems that nobody recently tried the input on the very first supporte
- sound card model, RK2. This patch fixes the byte offset to make i
- running again
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: snd_usb_caiaq: set mixernam
- alsamixer and friends want the mixername to be set. Even though th
- driver does not exports a real mixer device, export the name doesn'
- harm
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: snd_usb_caiaq: bump version numbe
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: snd_usb_caiaq: give better shortnam
- If not passed as module option, provide an own card ID with the newl
- introduced snd_set_card_id() call
- This will prevent ALSA from calling choose_default_name() which onl
- takes the last part of a name containing whitespaces. This for exampl
- caused 'Audio 4 DJ' to be shortened to 'DJ', which was not ver
- descriptive
- The implementation now takes the short name and removes all whitespace
- from it which is much nicer
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: snd_usb_caiaq: give better longnam
- The serial number is of no interest in the longname, remove it. Thi
- gives space for the usb path information which is more informative
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: snd_usb_caiaq: use strlcp
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: snd_usb_caiaq: clean whitespace
- Cosmetic changes only, no code change
- Signed-off-by: Daniel Mack <daniel@caiaq.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
USB generic driver
- - Fix usbmidi.patc
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - regenerate usbaudio.patc
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - ALSA: usb-audio - Fix types taken in min(
- Fix the compile warning due to different integer types used in min()
- sound/usb/usbaudio.c: In function 'init_substream_urbs'
- sound/usb/usbaudio.c:1087: warning: comparison of distinct pointer types lacks a cas
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: usb-audio: do not make URBs longer than sync packet interva
- Using more packets in one URB do avoid interrupts does not make sens
- when we have a sync pipe whose packets generate interrupts more often
- Therefore, limit the URB size to the synchronization packet interval
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: usb-audio - Volume control quirk for QuickCam E 350
- - E3500 report cval->max more than it actually can handel, so if yo
- set 95% capture level it will be silently muted
- - Betwen cval->min and cval-max(real) is 2940 control units
- but real are only 7 with cval->res = 384
- - Alsa can't handel less than 10 controls, so make it mor
- and set cval->res = 192
- Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: usb-audio: add MIDI drain callbac
- When draining, instead of waiting for fifty milliseconds, just wait fo
- the currently active URBs to complete. This cuts the usual waiting tim
- down to one USB frame, or zero in the common case when there is no URB
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: usb-audio: use multiple output URB
- Some newer USB MIDI interfaces use rather small packet sizes, so to ge
- enough bandwidth, we have to be able to send multiple packets in one US
- frame, so we have to use multiple URBs
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: usb-audio: use multiple input URB
- Some newer USB MIDI interfaces use rather small packet sizes, so to ge
- enough bandwidth, we have to be able to receive multiple packets in on
- USB frame, so we have to use multiple URBs
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: usb-audio: Xonar U1 digital output suppor
- Add support for the Asus Xonar U1. This device is mostly class compliant, bu
- the digital output requires a vendor-specific request
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: usb-audio: add workaround for Blue Microphones device
- Blue Microphones USB devices have an alternate setting that sends tw
- channels of data to the computer. Unfortunately, the descriptors o
- that altsetting have a wrong channel setting, which means that an
- recorded data from such a device has twice the sample rate from wha
- would be expected
- This patch adds a workaround to ignore that altsetting. Since thes
- devices have only one actual channel, no data is lost
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Cc: <stable@kernel.org
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: usb-audio - Correct bogus volume dB informatio
- Some USB devices give bogus dB information and it screws up PA
- It's better to detect a broken value and correct it in the drive
- before exposing the value to the outside
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: usb-audio - Use the new TLV_DB_MINMAX typ
- Use the new TLV_DB_MINMAX type instead of TLV_DB_SCALE
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: usb-audio - rework quirk for TerraTec Aureon USB 5.1 MkI
- This patch changes yet again the ID for the TTA cards, resulting in
- more reasonable name
- 1 [Aureon51MkII ]: USB-Audio - Aureon5.1MkI
- TerraTec Aureon5.1MkII at usb-0000:00:03.0-2, full spee
- Signed-off-by: Andrea Borgia <andrea@borgia.bo.it
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - trivial: remove extra spac
- Just for the sake of readability, removing extra spac
- Signed-off-by: Viral Mehta <viral.mehta@einfochips.com
- Signed-off-by: Jiri Kosina <jkosina@suse.cz
- - ALSA: usb - Add boot quirk for C-Media 6206 USB Audi
- Added boot quirk for C-Media CM6206 device in snd_usb_audio_probe
- The function snd_usb_cm6206_boot_quirk sets up six internal 16-bi
- registers in order to initialize the device. Values for the register
- came from sniffing USB traffic under Windows since only four of the si
- are documented in the datasheet for CM106 and some reserved bits wer
- also being set
- [Minor coding-style fixes by tiwai
- Signed-off-by: Dan Allongo <gongo2k1@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: usb-audio - errata corrige for quir
- Cut'n'paste mistake, whose likely result was nothing at all
- Correct version is "USB_DEVICE", not "USB_DEVICE_VENDOR_SPEC"
- Signed-off-by: Andrea Borgia <andrea@borgia.bo.it
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: usb-audio - Add quirk for Roland/Edirol M-16D
- Added a half-working quirk for Roland/Edirol M-16DX
- This enables the capture on the device but the playback on it seems stil
- problematic becuase of lack of sync with the capture clock
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: usb-audio - quirk for USB Aureon card
- Add quirk to provide proper naming of the Terratec Aureon 5.1 MkI
- USB card
- Signed-off-by: Andrea Borgia <andrea@borgia.bo.it
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: usbaudio - Add delay accoun
- Manage the PCM delay account based on the queued URBs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - sound: usb-audio: make the MotU Fastlane work agai
- Kernel 2.6.18 broke the MotU Fastlane, which uses duplicate endpoin
- numbers in a manner that is not only illegal but also confuses th
- kernel's endpoint descriptor caching mechanism. To work around this, w
- have to add a separate usb_set_interface() call to guide the USB core t
- the correct descriptors
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Reported-and-tested-by: David Fries <david@fries.net
- Cc: <stable@kernel.org
Utils
- - alsa-info: Version bump to 0.4.5
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info: use mktemp -
- Use mktemp -t instead of -p /tmp
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info: Check errors from mktem
- Check errors from mktemp
- Also remove superfluous mkdir $TEMPDIR. mktemp creates by itself
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info: revert the behavior of update optio
- It's not correct to invoke shell again for --update option
- Reverted to the old behavior, but to save without renaming to a fixe
- path for safety reasons
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info: Add --output optio
- Add --output option to specify the output file
- When invoked without --output option, alsa-info.sh now puts the fil
- into a temp file created by mktemp for security reasons
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info: Fix usage outpu
- Use tab to align the usage outputs
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info: Run the new update script automaticall
- Run the new updated script automatically without storing to the fixe
- path
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info: Use sysfs if available instead of dmidecod
- Using sysfs for acquiring DMI data requires no root privileges. Us
- it if available instead of dmidecode
- Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info.sh: include 1 line of dmesg contex
- So as to include possible ALSA messages without the common keywords
- as well as be more confident on the completeness/coverage of the report
- It also shows nice '--' separators between the dmesg ranges
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info.sh: add dmesg info on ALSA/HD
- Add outputs
- dmesg | grep -E 'ALSA|HDA|HDMI|sound|hda.codec|hda.intel
- which should cover most ALSA HDA kernel messages
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info: Version bump to 0.4.5
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info.sh: introduce withall(
- This merges duplicate code. The only behavior change is, we will now cal
- withsysfs() when no options are provided. I guess this is desired info
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info.sh: let mv fail loudl
- When mv cannot overwrite target file, it will prompt and return TRUE
- Add the '-f' option so that it returns FALSE when failed
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info.sh: fix whitespace leaked to stdou
- Redirect the "echo \t" outputs to the desired file
- and avoid messing up stdio
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info.sh: Do not automatically upload alsa inf
- - the greeting dialog informs that the script collects info, wait
- for OK button. It affords a concrete listing of information to collect
- /proc/asound/, aplay, etc. This not only shows respect for user privacy
- but also serves as basic debugging tips for ALSA newbies
- - when --upload option is given, the data will be automatically uploaded
- - when --no-upload option is given, the data is just stored locally and quit
- - when neither options are given, show a dialog to ask to upload or not
- The above ideas mostly come from Takashi
- Signed-off-by: Wu Fengguang <fengguang.wu@intel.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-info.sh: Provide system manufacturer and product name from DM
- This commit adds system manufacturer and product name information
- acquired using dmidecode to the output of the alsa-info script
- Note that those informations will only be available when dmidecod
- utility is installed and alsa-info is run with root privileges
- Signed-off-by: Ozan Çağlayan <ozan@pardus.org.tr
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add parsing of def_tristate to mod-dep
- The "def_tristate" is using in the recent Kconfig changes for th
- sequencer dependency clean-up
- Signed-off-by: Takashi Iwai <tiwai@suse.de
VIA82xx driver
- - ALSA: via82xx: add option to disable 500ms delay in snd_via82xx_codec_wai
- There's a large 500ms delay in snd_via82xx_codec_wait() that, at leas
- on my hardware, appears to be unnecessary. The rest of the init o
- the card works without logging any warnings or errors and both audi
- and mixer settings work
- This adds an "nodelay" parameter to disable this (undocumented in th
- code) large delay improving bootup time by 489-500ms
- [ 1.034217] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 505757 usec
- vs
- [ 0.533136] initcall alsa_card_via82xx_init+0x0/0x16 returned 0 after 15915 usec
- Signed-off-by: Simon Arlott <simon@fire.lp0.eu
- Signed-off-by: Takashi Iwai <tiwai@suse.de
Virtual Master
- - ALSA: Add new TLV types for dBwith min/ma
- Add new types for TLV dB scale specified with min/max values instea
- of min/step since the resolution can't match always with the on
- a device provides. For example, usb audio devices give 1/256 d
- resolution while ALSA TLV is based on 1/100 dB resolution
- The new min/max types have less problems because the possibl
- rounding error happens only at min/max
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
YMFPCI driver
- - sound: ymfpci: increase timer resolution to 96 kH
- Allow the interval timer to be programmed with its full 96 kH
- precision
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
au88x0 driver
- - sound: Use PCI_VDEVIC
- Signed-off-by: Joe Perches <joe@perches.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - ALSA: au88x0: fix wrong period_elapsed() cal
- The period_elapsed() call should be called when position moves
- The idea was taken from ALSA bug#4455
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - ALSA: au88x0: fix .pointer callbac
- Appearently, the used mask in the .pointer callback is invalid. It shoul
- be in period_bytes range. The period_bytes is pow(2), so simple bitwis
- operation is used
- Idea was taken from ALSA bug#4455
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
alsa-lib
Core
- - Release v1.0.2
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - add midi event test
- Add some tests for the snd_midi_event_* functions
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
Config API
- - fix doc error
- Fix various errors in the documentation that make doxygen complain
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - conf.c: more documentatio
- Expand the documentation for the snd_config_* functions
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
Control API
- - control.c: snd_ctl_wait: fix revents handlin
- The revents parameter of snd_ctl_poll_descriptors_revents() is a singl
- value, not an array
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - fix doc error
- Fix various errors in the documentation that make doxygen complain
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - Add the support of TLV_DB_MINMAX type
- Added the support of the new TLV_DB_MINMAX types
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Fix breakage of snd_card_load(
- Fixed the breakage of snd_card_load() for secondary and later card
- due to changes in snd_card_load1()
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - snd_card_get_index() - extend comment for last chang
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - Extend snd_card_get_index() to accept also control device name like /dev/snd/controlC
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Mixer API
- - remove unimplemented functions from header
- Remove some function declarations that are not (no longer) implemented
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
PCM API
- - pcm/ioplug: fix error code in start callbac
- When snd_pcm_start() is called in the invalid state, it should retur
- -EBADFD. But ioplug plugin returns -EAGAIN. Let's fix it
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pcm: workaround for avoiding automatic start in mmap mod
- In the normal mmap mode, the stream isn't started automatically even afte
- the data >= start_threshold has been written. However, in th
- mmap-emulation mode, the stream is started because it use
- snd_pcm_write_areas() internally
- As a workaround for this inconsistency, start_threshold value is change
- dynamically in sw_parmams and mmap_commit callbacks in mmap-emul plugin
- Meanwhile, start_threshold for slave PCM is set to boundary so that onl
- this plugin (or the one over it) can control the start of the stream
- This will fix problems in some apps using pulse plugin in the mmap mode
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - snd_pcm_scope_set_ops: make ops parameter cons
- The contents of the snd_pcm_scope_ops structure are not going to b
- changed, so we might as well declare is as constant. This change i
- backwards compatible, and avoids warnings if some ops structure i
- actually defined as const
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - Fix zero-division in pcm_rate.
- Patch from Debian bug#53945
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - remove unimplemented functions from header
- Remove some function declarations that are not (no longer) implemented
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - pcm_hooks: cosmetic removal of unused variable
- Signed-off-by: Paul Fertser <fercerpav@gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Manage dlobj lifetime in pcm_hooks.
- The shared object may be still needed depending on the implementatio
- of hook-installation functions. When any hooks are registered in th
- installation function, the dlobj has to be kept opened until closin
- the PCM instance
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pcm dmix plugin: fix MIX_AREAS_24 routine for i386 & x86_64 platform
- The code was copied from ALSA bug#4577 from CannibalZerg
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - Query the supported rate ranges from rate plugin
- Extend the PCM-rate plugin protocol to allow the host to query th
- supported sample rates. The protocol version is bumped to 0x010002
- and the version number negotiaion is slightly changed
- Now the plugin is supposed to fill the version it supports in return
- The old versioned plugins are still supported, but they may spe
- version-mismatch warning prints
- Signed-off-by: Takashi Iwai <tiwai@suse.de
RawMidi API
- - sound: rawmidi: disable active-sensing-on-close by defaul
- Sending an Active Sensing message when closing a port can interfere wit
- the following data if the port is reopened and a note-on is sent befor
- the device's timeout has elapsed. Therefore, it is better to disabl
- this setting by default
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
Sequencer API
- - more midi_event documentatio
- Expand the documentation for the snd_midi_event_* functions
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - seq_midi_event: fix decoding of (N)RPN event
- When decoding (N)RPN sequencer events into raw MIDI commands, th
- extra_decode_xrpn() function had accidentally swapped the MSB and LS
- controller values of both the parameter number and the data value
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - MIDI event decoder: prevent running status after syse
- Running status cannot be using in the command immediately followin
- a system exclusive command, so we have to reset the running statu
- state in that case
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
Timer API
- - timer_query: make ops structure constan
- The contents of the snd_timer_query_ops structure are not going to b
- changed, so we might as well declare is as constant. This change avoid
- a warning if some ops structure is actually defined as const
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
Configuration
- - Fix driver conf parsing in snd_config_hook_load_for_all_cards(
- Reported by Kevin Goodsell
- Summary: load_for_all_cards fails with existing configuration element
- In snd_config_hook_load_for_all_cards, the first call t
- snd_config_search attempts to locate an existing configuration node wit
- the name of the driver. Typically none is found, and everything i
- good. However, if such a node is located, the next line assumes it is
- leaf node with type 'string' and calls snd_config_get_string to fetc
- the string value. If this fails, the entire hook is abandoned
- Because of this, setting something like the following in asoundrc
- cards.<driver name>.foo
- is sufficient to disable the entire card-specific configuration
- As a concrete example, I have a HDA-Intel sound card. dmix.conf include
- a way to set period_size, period_time, and periods by usin
- configuration elements of the form cards.<driver name>.pcm.dmix.<var>
- In ~/.asoundrc I ad
- cards.HDA-Intel.pcm.dmix.period_size 102
- This will cause HDA-Intel.conf to fail to load, and the pcm defined i
- default.conf will fail to find the device-specific pc
- cards.HDA-Intel.pcm.default, and fall back on the default pcm usin
- plughw. By attempting to configure dmix, I have disabled it
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - conf.c: more documentatio
- Expand the documentation for the snd_config_* functions
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - conf.c: rename 'node' to 'config
- Just for consistency with the parameter names of all the othe
- functions
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - conf.c: rename 'leaf' to 'child
- Nodes that (might) have children are not leaves
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - conf.c: rename 'father' to 'parent
- I haven't found anything that would make compound nodes specificall
- male ..
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - conf.c: snd_config_add: prevent adopting a non-orpha
- When adding a configuration node to another, check that the child nod
- does not already have a parent. Otherwise, the old parent's childre
- list would become corrupted
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - USB-Audio.conf: fix definition for M-Audio AudioPhile spdif devic
- Add custom definitions for the AudioPhile "default" and "iec958" device
- so that output and input are routed to the correct PCM device
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - conf.c: fix handling of NULL string value
- Make sure that we do not crash when encountering configuration node
- with a NULL string value, or that at least we run into an assert()
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - conf.c: snd_config_set_id: prevent duplicate id
- snd_config_add() checks for duplicate ids, but it was possible to creat
- duplicates by adding a note and changing the id afterwards wit
- snd_config_set_id(); so we have to add a check there, too
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - conf.c: fix handling of NULL id
- Make sure that we do not crash when encountering configuration node
- with a NULL id. Furthermore, since we cannot avoid having NULL id
- anyway, allow the id of a top-level node to be reset to NULL
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - Fix SB-Xfi.con
- Added missing hint.device for rear, clfe, etc definitions
- Removed invalid iec958 capture definition
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add IEC958 status bits support to SB-XFi.con
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - Add config file for SB-XFi drive
- Signed-off-by: Takashi Iwai <tiwai@suse.de
Documentation
- - doc: hide structs with typedef
- In the documentation, hide structure types that have a correspondin
- typedef. Since doxygen 1.5.4, this is no longer the default whe
- OPTIMIZE_OUTPUT_FOR_C is set
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - doc: fix handling of @top_srcdir
- The value of top_srcdir should be replaced in the config file, not i
- the makefile, so we have to escape it in the makefile
- In the default case, the value of top_srcdir is ".." which, when used a
- a regular expression, is a little bit too inclusive
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
External PCM I/O Plugin SDK
- - fix doc error
- Fix various errors in the documentation that make doxygen complain
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
External Rate Converter Plugin SDK
- - Query the supported rate ranges from rate plugin
- Extend the PCM-rate plugin protocol to allow the host to query th
- supported sample rates. The protocol version is bumped to 0x010002
- and the version number negotiaion is slightly changed
- Now the plugin is supposed to fill the version it supports in return
- The old versioned plugins are still supported, but they may spe
- version-mismatch warning prints
- Signed-off-by: Takashi Iwai <tiwai@suse.de
I/O subsystem
- - fix doc error
- Fix various errors in the documentation that make doxygen complain
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
Test/Example code
- - add config test
- Add some test for the snd_config_* functions
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - test/lsb/midi_event.c: check for buffer size chec
- Add a test to check that snd_midi_event_decode() checks its outpu
- buffer size
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - test/lsb/midi_event.c: abort on fatal error
- If snd_midi_event_fails(), we cannot use the object and must abort th
- current test
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - test/pcm.c: float format suppor
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - add midi event test
- Add some tests for the snd_midi_event_* functions
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - test/pcm.c: Generic linear PCM suppor
- - Fix the support of non-native endiannes
- - Conversion for unsigned format
- - Only allow linear format
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - test/pcm.c: Fix S24 forma
- S24 format has different bit width and physical width
- For calculating the byte offset for big-endian packing, the latter valu
- has to be used
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - test/pcm.c: Sample generation on big endian platforms was broken
- Has not worked since commit 3d1fa924906996463ac33cba5b5143f762d913c
- Signed-off-by: Kenneth Johansson <kenneth@southpole.se
- Signed-off-by: Takashi Iwai <tiwai@suse.de
alsa-utils
Core
- - Release v1.0.2
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - alsamixer: show channel names for multichannel control
- For multichannel mixer controls, add the channel name to each scree
- control
- Also make some other small changes
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
/include/Makefile.am
- - alsamixer: show channel names for multichannel control
- For multichannel mixer controls, add the channel name to each scree
- control
- Also make some other small changes
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
ALSA Control (alsactl)
- - alsactl init rules: fix Lenovo T61 initialization (Speaker Playback Switch
- See: https://bugzilla.redhat.com/show_bug.cgi?id=50626
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - alsactl: init - fix default configuration for ENS137
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - alsactl: fixed Headphone Playback Volume setting in default rule
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Speaker Test
- - speaker-test: only check byte order onc
- Rather than having numerous preprocessor directives scattered in the cod
- checking __BYTE_ORDER, only check it once and define a set of macro
- accordingly that can be used in the rest of the code. This makes thing
- simpler to read and less error-prone
- Signed-off-by: Dan McGee <dpmcgee@gmail.com
- - speaker-test: move existing endian macros up in the fil
- This is necessary for a later patch removing the various endianness check
- sprinkled throughout the code
- Signed-off-by: Dan McGee <dpmcgee@gmail.com
- - Remove dead/commented out cod
- Signed-off-by: Dan McGee <dpmcgee@gmail.com
- - Allow frequencies down to 30 H
- Signed-off-by: Dan McGee <dpmcgee@gmail.com
- - speaker-test: allow frequency to be floating poin
- Use atof() rather than atoi() to store the frequency- we were already usin
- a floating point value internally but did not let the user specify one fro
- the command line
- Signed-off-by: Dan McGee <dpmcgee@gmail.com
alsamixer
- - alsamixer: fix display of inactive volume ba
- Fix the volume bar color selection logic so that the current attribut
- is used for inactive controls
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - alsamixer: rename attr to c
- Rename the attr variable because it contains not only the character'
- attributes but also the character itself
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - alsamixer - Tricolorize volume bar
- A little of bit of Italian taste was missing..
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsamixer: update man pag
- Update man page for change in "CAPTURE" field
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - alsamixer: fix text box clipping with multi-column character
- When a multi-column character would straddle the left window border o
- a text box, we have to take the inserted space character into accoun
- when we compute how many characters fit into the rest of the line
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
- - alsamixer - Fix uninitialized variable warnin
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsamixer: show channel names for multichannel control
- For multichannel mixer controls, add the channel name to each scree
- control
- Also make some other small changes
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
aplaymidi/arecordmidi
- - aplaymidi: reduce bandwidth for big SysEx message
- When throttling the data rate for big SysEx messages, use the bandwidt
- that devices use in practice instead of the theoretical maximum
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
alsa-tools
Core
- - Release v1.0.2
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
Envy24 Control
- - envy24control - Don't redeclare isblank()
- While technically isblank() is a C library function, nothing stops it fro
- being a macro, and indeed it seems to be on glibc-2.10
- This should not be a problem because ctype.h already declares it o
- probably all the systems where it's used
- Signed-off-by: Takashi Iwai <tiwai@suse.de
ac3dec (Dolby Digital Decoder)
- - ac3dec - Fix typos of -q optio
- It's quiet, not quit
- Signed-off-by: Takashi Iwai <tiwai@suse.de
hdspconf
- - Also fix the configure for hdspconf for LIBS/LDFLAGS mistakes
- Commit 56970e8143b4d171a118d114b1ddfa7621401127 already took care of thi
- for the other tools, but hdspconf somewhat was excluded, fix this now
- Signed-off-by: Takashi Iwai <tiwai@suse.de
qlo10k1
- - qlo10k1: Fix usage of $x_libraries in acinclude.m4 - it may be empt
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
us428control
- - us428control - Fix array overflo
- Fix the array overflow in accessing Vol[]
- Cus428State.cc:257:32: warning: array subscript is above array bound
- Signed-off-by: Takashi Iwai <tiwai@suse.de
alsa-plugins
Core
- - Release v1.0.2
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - pulse: use PA_CONTEXT_IS_GOOD where applicabl
- PA_CONTEXT_IS_GOOD is a safer way to check whether a context is stil
- valid
- This patch also bumps the version requirement of libpulse to 0.9.11
- Signed-off-by: Takashi Iwai <tiwai@suse.de
Documentation
- - speex - Add echo-cancelling option to speexdsp plugi
- Signed-off-by: Takashi Iwai <tiwai@suse.de
OSS Mixer -> ALSA Control plugin
- - oss - Add missing initialization of fragment
- The periods calculation was missing for initializing OSS fragments
- Signed-off-by: Takashi Iwai <tiwai@suse.de
Public Parrot Hack rate converter
- - Add PCM rates query support for PCM rate plugin
- Follow the new PCM rate-plugin protocol to support the rate rang
- queries, etc
- Signed-off-by: Takashi Iwai <tiwai@suse.de
PulseAudio -> ALSA plugin
- - pulse: immediately trigger EIO when connection is droppe
- When the connection is dropped notify the application immediatel
- instead of waiting until the applications calls into us the next time
- This makes "aplay" handle connections shutdown similar to hardwar
- unplugs: an immediate EIO is thrown
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pulse: rework object destruction paths a bi
- Make sure we deal better with partially initialized structs
- Don't check for pointer state before calling free() since free() doe
- that anyway
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pulse: unify stream/context state check
- Unify (and simplify) the paths that check for the validity of
- stream/context: always call into check_stream()/pulse_check_connection(
- when applicable instead of rolling our own checks each time. Also chec
- for validity of mainloop before locking it
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pulse: get rid of redundant state variabl
- snd_pulse_t::state was mostly shadowing the state o
- pa_context_get_state(snd_pulse_t::context), so get rid of it and use th
- state of the context directly
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pulse: move a couple of PCM related functions from pulse.c to pcm_pulse.
- A number of functions in pulse.c are only relevant for the PCM handling
- so let's move them to pcm_pulse.c. This allows us to simplify thei
- argument lists a bit
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pulse: replace manual mainloop by pa_mainloop_iterate(
- The pa_mainloop_prepare()/_poll()/_dispatch() can be simplified b
- simply calling pa_mainloop_iterate() which does all this in one call
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pulse: call pa_threaded_mainloop_wait() to handle spurious wakeup
- pa_threaded_mainloop_wait() can wake up for no reason, according to th
- specs of the underlying POSIX ptrhead_cond_wait() docs, so we need t
- call it in a loop here which should be cleaner anyway
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pulse: unify destruction of snd_pulse_
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pulse: use PA_CONTEXT_IS_GOOD where applicabl
- PA_CONTEXT_IS_GOOD is a safer way to check whether a context is stil
- valid
- This patch also bumps the version requirement of libpulse to 0.9.11
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - pulse: get rid of a number of assert()
- Instead of hitting an assert when any of the plugin functions is calle
- in an invalid context we should return a clean error to make sur
- programs are not unnecessarily aborted
- This should fix issues such as http://pulseaudio.org/ticket/59
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsa-plugins/pulse: Implement 'pause'
- Just cork or uncork the stream to pause or unpause it
- Signed-off-by: Troy Moure <twmoure@szypr.net
- Signed-off-by: Takashi Iwai <tiwai@suse.de
Speex PCM plugin
- - speex - Add echo-cancelling option to speexdsp plugi
- Signed-off-by: Takashi Iwai <tiwai@suse.de
libavcodec's resampler
- - Add PCM rates query support for PCM rate plugin
- Follow the new PCM rate-plugin protocol to support the rate rang
- queries, etc
- Signed-off-by: Takashi Iwai <tiwai@suse.de
alsa-python
Core
- - Release v1.0.2
- Signed-off-by: Jaroslav Kysela <perex@perex.cz
- - [PATCH] alsa-python: Add support for setuptool
- This patch adds support for setuptools to the setup.py file of python-alsa
- Signed-off-by: Christopher Arndt <chris@chrisarndt.de
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
pyalsa.alsaseq module
- - pyalsa: fix integer overflow in alsaseq.
- * Original patch description
- I've been using the alsaseq python module and I found a bug. Sometime
- the SEQ_* constants have extremely large and incorrect values. Fo
- example, 25769803811 instead of 35. The lower 32-bits are alway
- correct
- Obviously, I'm running a 64-bit operating system
- The problem is that the `value` member of the `ConstantObject
- structure is an `unsigned int` whereas it should be a `long`. I'v
- attached a patch. It's against the latest released version, 1.0.20
- * Revised patch description
- I just noticed that `PyString_FromFormat` in Python 2.6 doesn't handl
- `%lx` in format strings, so my patch breaks `repr(Constant)`
- In addition to that, it was not necessary to change the signedness of `value`
- With those two things in mind, the patch perhaps ought to look lik
- this (see attached)
- Signed-off-by: Chris Coleman <chris.coleman83 at gmail.com
- Signed-off-by: Takashi Iwai <tiwai@suse.de
- - alsaseq: fix time stamp
- The number of nanoseconds per second is actually 10^9
- Signed-off-by: Clemens Ladisch <clemens@ladisch.de
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